Claude Hayn
2011-May-17 01:27 UTC
[asterisk-users] 3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33
Alex,
Thank you so much for your response. I've been so consumed with other
business that I only just now getting back to this issue. We have
implemented your suggestion which is perfect. Thank you again.
I've never asked a question of the community before and I'm extremely
happy
with the rapid response I received.
Somewhat related to this initial problem I have an additional
problem.
In extensions.conf we have identified one ITSP for sending outbound calls
to. From the problems we've been having with our IPSPs being unavailable at
different times we need to be able to send calls to ITSPs that are available
in a cascading fashion.
I would like to know how to configure extensions.conf so that calls will
attempt in the cascading fashion sending calls to our our ITSPs until one
excepts the call.
I was thinking of implementing something like this, but wonder if I should
add some sort of timer or delay? I would like to call to the excepted by
the first available ITSP in some order.
; send all outgoing calls directly to the ITSP
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@ITSP-one)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@ITSP-two)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@ITSP-three)
Thank you for taking a look,
Claude
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
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asterisk-users-request at lists.digium.com
Sent: Tuesday, May 10, 2011 3:20 AM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 82, Issue 33
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Today's Topics:
1. Re: QueueCallerAbandon is not triggering after 1.8.3.3...
(Louis Carreiro)
2. ITSP Multi IPs (Claude Hayn)
3. Re: ITSP Multi IPs (Alex Balashov)
4. 1.8 and prematuremedia problem (d tbsky)
5. Re: 1.8 and prematuremedia problem (Satish Patel)
6. Re: 1.8 and prematuremedia problem (d tbsky)
7. Re: Background music during a call (Rizwan Hisham)
8. Re: OT - Which Android handset with Wifi-only ? (Olivier)
9. Re: OUTBOUND CALLER ID (mahesh katta)
10. Re: OUTBOUND CALLER ID (DHAVAL INDRODIYA)
11. Re: OUTBOUND CALLER ID (mahesh katta)
12. Re: 40sec between dial execution and sending SIP request
(Pezhman Lali)
----------------------------------------------------------------------
Message: 1
Date: Mon, 9 May 2011 20:54:39 -0400
From: Louis Carreiro <carreirolt at gmail.com>
Subject: Re: [asterisk-users] QueueCallerAbandon is not triggering
after 1.8.3.3...
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <BANLkTi=H24L6Mu-fxN0Cc-fbpqGXFRJYww at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Has anyone else noticed this?
v/r,
Me
On Fri, May 6, 2011 at 12:11 PM, Louis Carreiro <carreirolt at
gmail.com>wrote:
> Has anyone else noticed that QueueCallerAbandon is not showing up in
> the AMI after the 1.8.3.3? Am I missing something? I'm getting what
> seems like everything else but QueueCallerAbandon.
>
> v/r,
> Me
>
>
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Message: 2
Date: Mon, 9 May 2011 21:12:53 -0400
From: "Claude Hayn" <chayn123 at gmail.com>
Subject: [asterisk-users] ITSP Multi IPs
To: <asterisk-users at lists.digium.com>
Message-ID: <00c601cc0eaf$642a4820$2c7ed860$@gmail.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I'm hoping someone has a suggestion for us.
We have an ITSP that sends inbound traffic to us. Unannounced to us last
week they started alternately sending traffic from two IP addresses, instead
of the one we knew about. Some calls would pass, and others would be dumped
as unauthenticated.
I added the 2nd IP to the sip.conf file to allow for this, and everything
was fine until this morning.
This morning the first IP started being rejected even though it was listed
in the sip.conf file. As soon as I commented out the .45 IP address that I
added last week traffic flowed again.
We need to authorize traffic from both IP addresses.
Any suggestions?
Sip.conf:
[incoming-trunk]
type=peer
insecure=port,invite
context=default-incoming
host=XX.XXX.XXX.40
canreinvite=nonat
qualify=yes
disallow=all
allow=ulaw
nat=yes
[incoming-trunk]
type=peer
insecure=port,invite
context=default-incoming
host=XX.XXX.XXX.45
canreinvite=nonat
qualify=yes
disallow=all
allow=ulaw
nat=yes
Default-incoming in extensions.conf is:
[default-incoming]
; Create extensions 1234 & 1235 for evaluating echo latency.
exten => 1234,1,Playback(demo-echotest) ; Let them know what's
going
on
exten => 1234,n,Playback(beep) ; Beep to let them know the test has
started
exten => 1234,n,Echo ; Do the echo test
exten => 1234,n,Playback(demo-echodone) ; Let them know it's over
exten => 1234,n,Goto(s,6) ; Start over
; send a 1KHz test tone
exten => 1235,1,MilliWatt
; play the hold music
exten => 1236,1,Answer
exten => 1236,n,MusicOnHold(default)
exten => _X.,1,Dial(DAHDI/R0/${EXTEN:4})
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Message: 3
Date: Mon, 09 May 2011 21:16:43 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [asterisk-users] ITSP Multi IPs
To: asterisk-users at lists.digium.com
Message-ID: <4DC891FB.4090805 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
You need to give the two sip.conf peers different names (in square
brackets).
On 05/09/2011 09:12 PM, Claude Hayn wrote:
> Hi,
>
> I'm hoping someone has a suggestion for us.
>
> We have an ITSP that sends inbound traffic to us. Unannounced to us
> last week they started alternately sending traffic from two IP
> addresses, instead of the one we knew about. Some calls would pass,
> and others would be dumped as unauthenticated.
>
> I added the 2^nd IP to the sip.conf file to allow for this, and
> everything was fine until this morning.
>
> This morning the first IP started being rejected even though it was
> listed in the sip.conf file. As soon as I commented out the .45 IP
> address that I added last week traffic flowed again.
>
> We need to authorize traffic from both IP addresses.
>
> Any suggestions?
>
> Sip.conf:
>
> [incoming-trunk]
>
> type=peer
>
> insecure=port,invite
>
> context=default-incoming
>
> host=XX.XXX.XXX.40
>
> canreinvite=nonat
>
> qualify=yes
>
> disallow=all
>
> allow=ulaw
>
> nat=yes
>
> [incoming-trunk]
>
> type=peer
>
> insecure=port,invite
>
> context=default-incoming
>
> host=XX.XXX.XXX.45
>
> canreinvite=nonat
>
> qualify=yes
>
> disallow=all
>
> allow=ulaw
>
> nat=yes
>
> Default-incoming in extensions.conf is:
>
> [default-incoming]
>
> ; Create extensions 1234 & 1235 for evaluating echo latency.
>
> exten => 1234,1,Playback(demo-echotest) ; Let them know what's going
> on
>
> exten => 1234,n,Playback(beep) ; Beep to let them know the test has
> started
>
> exten => 1234,n,Echo ; Do the echo test
>
> exten => 1234,n,Playback(demo-echodone) ; Let them know it's over
>
> exten => 1234,n,Goto(s,6) ; Start over
>
> ; send a 1KHz test tone
>
> exten => 1235,1,MilliWatt
>
> ; play the hold music
>
> exten => 1236,1,Answer
>
> exten => 1236,n,MusicOnHold(default)
>
> exten => _X.,1,Dial(DAHDI/R0/${EXTEN:4})
>
>
>
> --
> _____________________________________________________________________
> -- 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
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
------------------------------
Message: 4
Date: Tue, 10 May 2011 09:57:56 +0800
From: d tbsky <tbskyd at gmail.com>
Subject: [asterisk-users] 1.8 and prematuremedia problem
To: asterisk-users at lists.digium.com
Message-ID: <BANLkTi=Jqa0wpjYEvbso9xjX_RCwmjzdgA at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
hi:
our current connection is below:
sip phone<--->asterisk<---->alcatel PBX<---->PSTN
asterisk and alcatel PBX is connected via E1 isdn-pri.
when I use sip phone to dial outside PSTN world:
1. with 1.4 it is fine.
2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip
phone can not hear the ring and the beginning of the PSTN voice.
3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN
voice. I try to play options with "prematuremedia" and
"progressinband". but I can not find working settings.
I don't know what other options I can try.
thank a lot for information!!
------------------------------
Message: 5
Date: Mon, 9 May 2011 22:42:51 -0400
From: Satish Patel <satish_lx at hotmail.com>
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BLU0-SMTP19808D6603D197E5D609B1A90870 at phx.gbl>
Content-Type: text/plain; charset="us-ascii"; format=flowed
Apply this patch https://issues.asterisk.org/view.php?id=18868
--
Sent from my iPhone
On May 9, 2011, at 9:57 PM, d tbsky <tbskyd at gmail.com> wrote:
> hi:
> our current connection is below:
>
> sip phone<--->asterisk<---->alcatel PBX<---->PSTN
>
> asterisk and alcatel PBX is connected via E1 isdn-pri.
>
> when I use sip phone to dial outside PSTN world:
> 1. with 1.4 it is fine.
> 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip
> phone can not hear the ring and the beginning of the PSTN voice.
> 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN
> voice. I try to play options with "prematuremedia" and
> "progressinband". but I can not find working settings.
>
> I don't know what other options I can try.
> thank a lot for information!!
>
> --
> _____________________________________________________________________
> -- 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
>
------------------------------
Message: 6
Date: Tue, 10 May 2011 10:57:02 +0800
From: d tbsky <tbskyd at gmail.com>
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTimXv9AdrmG_tDH4y=7RG=j5Fxa39A at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
hi:
thanks a lot for your quick reply. I saw that patch and think that
it was already included in 1.8.3.
now I know it will be included in 1.8.5.
I will try it and thanks again for your kindly help!!
2011/5/10 Satish Patel <satish_lx at hotmail.com>:> Apply this patch https://issues.asterisk.org/view.php?id=18868
>
> --
> Sent from my iPhone
>
> On May 9, 2011, at 9:57 PM, d tbsky <tbskyd at gmail.com> wrote:
>
>> hi:
>> ? our current connection is below:
>>
>> ? sip phone<--->asterisk<---->alcatel PBX<---->PSTN
>>
>> ?asterisk and alcatel PBX is connected via ?E1 isdn-pri.
>>
>> ?when I ?use sip phone to dial outside PSTN world:
>> ?1. with 1.4 it is fine.
>> ?2. with 1.6.2, I need to set prematuremedia=no is sip.conf. ?or sip
>> phone can not hear the ring and the beginning of the PSTN voice.
>> ?3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN
>> voice. I try to play options with "prematuremedia" and
>> "progressinband". but I can not find working settings.
>>
>> ?I don't know what other options I can try.
>> ?thank a lot for information!!
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
------------------------------
Message: 7
Date: Tue, 10 May 2011 08:33:17 +0500
From: Rizwan Hisham <rizwanhasham at gmail.com>
Subject: Re: [asterisk-users] Background music during a call
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTi=6-8X2bbDTQUU_VfeJ_3+7O7MzWg at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanks a lot loan. Will try it today.
Cheers
On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias <indreias at gmail.com>
wrote:
> Updated dialplan: fix a typo when using MOH variable and now you have
> truly dynamic conference rooms.
>
> Have fun,
> Ioan.
>
> +++++++++++++++++++++++++++++++++++++++++
> exten => _[12]XXX,1,Set(__MM=${EPOCH})
> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
>
> [chat-room]
> exten => love,1,Goto(love-a,1)
> exten => love,2,Goto(love-b,1)
>
> exten => love-a,1,Set(__MOH=love)
> exten => love-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
>
> exten => love-b,1,Goto(chat,100)
>
> exten => curse,1,Goto(curse-a,1)
> exten => curse,2,Goto(curse-b,1)
>
> exten => curse-a,1,Set(__MOH=curse)
> exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
>
> exten => curse-b,1,Goto(chat,100)
>
> exten => fake,1,Answer
> exten => fake,2,MusicOnHold(${MOH})
>
> exten => chat,1,Goto(100)
> exten => chat,2,MeetMe(${MM},dx1q)
>
> exten => chat,100,MeetMe(${MM},daAx1q)
>
> exten => h,1,MeetMeAdmin(${MM},K)
> +++++++++++++++++++++++++++++++++++++++++
>
> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias <indreias at gmail.com>
wrote:
> > I have tested the following dialplan and it could be used as a
> > starting point. What you have to resolve is how to generate different
> > MeetMe conference room - in the example we have only one room = 1234
> >
> > If you prefix the dialled extension with 1 => you will have a
"lovely
> > chat". With 2 -> "cursing chat".
> >
> > HTH,
> >
> > Ioan
>
> --
> _____________________________________________________________________
> -- 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
>
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 3333 6767 26
E: rizwanhasham at gmail.com
W: www.axvoice.com
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Message: 8
Date: Tue, 10 May 2011 06:52:14 +0200
From: Olivier <oza_4h07 at yahoo.fr>
Subject: Re: [asterisk-users] OT - Which Android handset with
Wifi-only ?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTikHQKjE6XCpnJEHoBkCFMX59TVyYw at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2011/5/9 Olivier <oza_4h07 at yahoo.fr>
>
> 2011/5/9 randulo <randulo at randulo.com>
>
>> On Mon, May 9, 2011 at 2:20 PM, <mgraves at mstvp.com> wrote:
>> > Lots of Android handsets support wifi, like my G2, aka HTC
DesireZ.
>>
>> Wouldn't ANY modern one have wifi? That would be odd if it
didn't, would
>> it not?
>>
>>
> Yes, of course, all dual-mode phones support WiFi but :
> 1. I'm not certain those would work without any SIM-card inside
> 2. those are likely to be more expensive than WiFi-only handset.
>
> See the last iPod touch which is marketed as a Sametime
>
I meant "Face Time" (I should n't (have stopped to) drink while
typing ;-))
client is quite cheeper than the iPhone.>
> To my knowledge, most Android-based WiFi-only machines are tablets.
>
> Cheers
>
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Message: 9
Date: Tue, 10 May 2011 10:45:15 +0530
From: mahesh katta <maheshkatta at flexydial.com>
Subject: Re: [asterisk-users] OUTBOUND CALLER ID
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTimaDHrvChXf3-6rhiXq2J4xWeCq+Q at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
sir,
Below configuration i wase made in server . but this is not working.
exten => _90XXXXXXXXX,1,NoOp(${CALLERID(num)})
exten => _90XXXXXXXXX/5001,2,Set(CALLERID(name)=44578999)
exten => _90XXXXXXXXX,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _90XXXXXXXXX/5001,4,Set(CALLERID(num)=44578999)
exten =>
_90XXXXXXXXX,5,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERID
NUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten => _90XXXXXXXXX,6,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten => _90XXXXXXXXX,7,Hangup
On Mon, May 9, 2011 at 8:14 PM, Carlos Rojas <crt.rojas at gmail.com>
wrote:
> Hello
>
> Do you set your callerid in the context outgoing?
>
> [outgoing]
>
> exten => _X.,1,Set(CALLERID(num)=4663000)
> exten => _X.,n,Dial(..............
>
> On Mon, May 9, 2011 at 4:45 AM, mahesh katta
<maheshkatta at flexydial.com>wrote:>
>> Sir ,
>>
>> this is not working
>>
>>
>> On Mon, May 9, 2011 at 1:52 PM, A J Stiles <asterisk_list at
earthshod.co.uk
>> > wrote:
>>
>>> On Monday 09 May 2011, mahesh katta wrote:
>>> > Hi,
>>> > THIS IS IN DUBAI.
>>> >
>>> > I am having PRI line with 100 DID's (00-99) and when we
call to any
>>> > landline or mobile number then it shows us our board number or
pilot
>>> number
>>> > (i.e 4663000 means 00)..
>>>
>>> In the context through which outgoing calls are placed, you need a
step
>>> which
>>> sets the caller ID number. For instance, part of our dialplan maps
>>> external
>>> phone numbers with the local part 707060 to 707072 to internal
extensions>>> 301
>>> to 312 respectively. Our E1 provider also requires us to include
the
STD>>> code, minus the leading zero, for the town we are in -- and will
silently>>> anonymise the call if we try to send a caller ID that does not
belong to
>>> us.
>>>
>>> So for outgoing calls, we have something like
>>>
>>> [ts-outgoing]
>>> exten => _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
>>> exten => _0., 2, Set(CALLERID(num)=${STD}${localno})
>>>
>>>
>>> --
>>> AJS
>>>
>>> Answers come *after* questions.
>>>
>>> --
>>>
_____________________________________________________________________
>>> -- 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
>>>
>>
>>
>>
>> --
>> Best Regards,
>>
>> Mahesh Katta
>> *BUZZ**WORKS* Business Services Private Limited
>> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
>> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road,
Andheri
>> (E) Mumbai 400069
>> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
>> Web http://www.buzzworks.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
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Message: 10
Date: Tue, 10 May 2011 12:00:54 +0530
From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
Subject: Re: [asterisk-users] OUTBOUND CALLER ID
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTimEPoWxVe0Tu4NoSsdZt3RO=o1n3w at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Mahesh,
I have solutions but its paid, i can provide setting if you are interested
on counsultancy
Cheers
Dhaval
On Tue, May 10, 2011 at 10:45 AM, mahesh katta
<maheshkatta at flexydial.com>wrote:
> sir,
>
> Below configuration i wase made in server . but this is not working.
>
>
> exten => _90XXXXXXXXX,1,NoOp(${CALLERID(num)})
> exten => _90XXXXXXXXX/5001,2,Set(CALLERID(name)=44578999)
> exten => _90XXXXXXXXX,3,AGI(agi://127.0.0.1:4577/call_log)
> exten => _90XXXXXXXXX/5001,4,Set(CALLERID(num)=44578999)
> exten =>
>
_90XXXXXXXXX,5,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERID
NUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))> exten => _90XXXXXXXXX,6,Dial(${TRUNK}/${EXTEN:1},,tTo)
> exten => _90XXXXXXXXX,7,Hangup
>
>
>
> On Mon, May 9, 2011 at 8:14 PM, Carlos Rojas <crt.rojas at gmail.com>
wrote:
>
>> Hello
>>
>> Do you set your callerid in the context outgoing?
>>
>> [outgoing]
>>
>> exten => _X.,1,Set(CALLERID(num)=4663000)
>> exten => _X.,n,Dial(..............
>>
>> On Mon, May 9, 2011 at 4:45 AM, mahesh katta
<maheshkatta at flexydial.com>wrote:>>
>>> Sir ,
>>>
>>> this is not working
>>>
>>>
>>> On Mon, May 9, 2011 at 1:52 PM, A J Stiles <
>>> asterisk_list at earthshod.co.uk> wrote:
>>>
>>>> On Monday 09 May 2011, mahesh katta wrote:
>>>> > Hi,
>>>> > THIS IS IN DUBAI.
>>>> >
>>>> > I am having PRI line with 100 DID's (00-99) and when
we call to any
>>>> > landline or mobile number then it shows us our board
number or pilot
>>>> number
>>>> > (i.e 4663000 means 00)..
>>>>
>>>> In the context through which outgoing calls are placed, you
need a step
>>>> which
>>>> sets the caller ID number. For instance, part of our dialplan
maps
>>>> external
>>>> phone numbers with the local part 707060 to 707072 to internal
>>>> extensions 301
>>>> to 312 respectively. Our E1 provider also requires us to
include the
>>>> STD
>>>> code, minus the leading zero, for the town we are in -- and
will
>>>> silently
>>>> anonymise the call if we try to send a caller ID that does not
belong
to>>>> us.
>>>>
>>>> So for outgoing calls, we have something like
>>>>
>>>> [ts-outgoing]
>>>> exten => _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
>>>> exten => _0., 2, Set(CALLERID(num)=${STD}${localno})
>>>>
>>>>
>>>> --
>>>> AJS
>>>>
>>>> Answers come *after* questions.
>>>>
>>>> --
>>>>
_____________________________________________________________________
>>>> -- 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
>>>>
>>>
>>>
>>>
>>> --
>>> Best Regards,
>>>
>>> Mahesh Katta
>>> *BUZZ**WORKS* Business Services Private Limited
>>> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
>>> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road,
Andheri
>>> (E) Mumbai 400069
>>> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
>>> Web http://www.buzzworks.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
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>
>
>
> --
> Best Regards,
>
> Mahesh Katta
> *BUZZ**WORKS* Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E) Mumbai 400069
> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Web http://www.buzzworks.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
>
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------------------------------
Message: 11
Date: Tue, 10 May 2011 12:03:47 +0530
From: mahesh katta <maheshkatta at flexydial.com>
Subject: Re: [asterisk-users] OUTBOUND CALLER ID
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTinV5Ow0cKp2M7mpqbLPytmC_Kx-AQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Daval,
how much you require for this .
On Tue, May 10, 2011 at 12:00 PM, DHAVAL INDRODIYA <dhaval.it01034 at
gmail.com> wrote:
> Hi Mahesh,
>
> I have solutions but its paid, i can provide setting if you are
> interested on counsultancy
>
> Cheers
> Dhaval
>
>
> On Tue, May 10, 2011 at 10:45 AM, mahesh katta
<maheshkatta at flexydial.com>wrote:>
>> sir,
>>
>> Below configuration i wase made in server . but this is not working.
>>
>>
>> exten => _90XXXXXXXXX,1,NoOp(${CALLERID(num)})
>> exten => _90XXXXXXXXX/5001,2,Set(CALLERID(name)=44578999)
>> exten => _90XXXXXXXXX,3,AGI(agi://127.0.0.1:4577/call_log)
>> exten => _90XXXXXXXXX/5001,4,Set(CALLERID(num)=44578999)
>> exten =>
>>
_90XXXXXXXXX,5,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERID
NUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))>> exten => _90XXXXXXXXX,6,Dial(${TRUNK}/${EXTEN:1},,tTo)
>> exten => _90XXXXXXXXX,7,Hangup
>>
>>
>>
>> On Mon, May 9, 2011 at 8:14 PM, Carlos Rojas <crt.rojas at
gmail.com> wrote:
>>
>>> Hello
>>>
>>> Do you set your callerid in the context outgoing?
>>>
>>> [outgoing]
>>>
>>> exten => _X.,1,Set(CALLERID(num)=4663000)
>>> exten => _X.,n,Dial(..............
>>>
>>> On Mon, May 9, 2011 at 4:45 AM, mahesh katta
<maheshkatta at flexydial.com>wrote:>>>
>>>> Sir ,
>>>>
>>>> this is not working
>>>>
>>>>
>>>> On Mon, May 9, 2011 at 1:52 PM, A J Stiles <
>>>> asterisk_list at earthshod.co.uk> wrote:
>>>>
>>>>> On Monday 09 May 2011, mahesh katta wrote:
>>>>> > Hi,
>>>>> > THIS IS IN DUBAI.
>>>>> >
>>>>> > I am having PRI line with 100 DID's (00-99) and
when we call to any
>>>>> > landline or mobile number then it shows us our board
number or pilot
>>>>> number
>>>>> > (i.e 4663000 means 00)..
>>>>>
>>>>> In the context through which outgoing calls are placed, you
need a
step>>>>> which
>>>>> sets the caller ID number. For instance, part of our
dialplan maps
>>>>> external
>>>>> phone numbers with the local part 707060 to 707072 to
internal
>>>>> extensions 301
>>>>> to 312 respectively. Our E1 provider also requires us to
include the
>>>>> STD
>>>>> code, minus the leading zero, for the town we are in -- and
will
>>>>> silently
>>>>> anonymise the call if we try to send a caller ID that does
not belong
>>>>> to us.
>>>>>
>>>>> So for outgoing calls, we have something like
>>>>>
>>>>> [ts-outgoing]
>>>>> exten => _0., 1,
Set(localno=7070$[${CALLERID(num)}-240])
>>>>> exten => _0., 2, Set(CALLERID(num)=${STD}${localno})
>>>>>
>>>>>
>>>>> --
>>>>> AJS
>>>>>
>>>>> Answers come *after* questions.
>>>>>
>>>>> --
>>>>>
_____________________________________________________________________
>>>>> -- 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Best Regards,
>>>>
>>>> Mahesh Katta
>>>> *BUZZ**WORKS* Business Services Private Limited
>>>> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
>>>> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla
Road,
Andheri>>>> (E) Mumbai 400069
>>>> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax
+91.22.4229.2634
>>>> Web http://www.buzzworks.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
>>>>
>>>
>>>
>>> --
>>>
_____________________________________________________________________
>>> -- 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
>>>
>>
>>
>>
>> --
>> Best Regards,
>>
>> Mahesh Katta
>> *BUZZ**WORKS* Business Services Private Limited
>> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
>> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road,
Andheri
>> (E) Mumbai 400069
>> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
>> Web http://www.buzzworks.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
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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------------------------------
Message: 12
Date: Tue, 10 May 2011 11:48:11 +0430
From: Pezhman Lali <lopl at lopl.net>
Subject: Re: [asterisk-users] 40sec between dial execution and sending
SIP request
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTi=VJdo-wwuT36PNMRoc-1uJ30F7XQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
thanks,
this delay is occurred on asterisk server, between dial execution and
"CALLED ....."
On Mon, May 9, 2011 at 7:12 PM, Warren Selby <wcselby at selbytech.com>
wrote:
> On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali <lopl at lopl.net>
wrote:
>
>> Dear
>> I have a small pbx with asterisk 1.6.2.16.
>> I have a funny problem, there is exactly 40sec between dial execution
and
>> sending first invite packet on sip.
>> do you have any idea where the problem is ?
>>
>
> Check the dial timeout on your phone itself. What model phone do you
have?>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.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
>
--
Pezhman Lali
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