[asterisk-users] Asterisk 1.8 not playing parking slot announcement to parker

2012-10-25 Thread John Taylor
Just upgraded to 1.8, we use the multi lot parking feature by dialling *4.
We are not getting the parking slot announcement being played to the person
who parks the call, so it's impossible to tell which slot they've gone
into. Could someone check our config?

On Debian Squeeze using packages from
http://packages.asterisk.org/debsqueeze main (Asterisk
1.8.11.1-1digium1~squeeze)

/etc/asterisk/features.conf
[general]
transferdigittimeout = 5 ; Number of seconds to wait between digits when
transferring a call
xfersound = beep ; to indicate an attended transfer is complete
xferfailsound = unavailablebeep ; to indicate a failed transfer
featuredigittimeout = 2000 ; Max time (ms) between digits for
atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer
default is 15 seconds.
[featuremap]
atxfer = *1 ; Attended transfer
blindxfer = *2 ; Blind transfer  (default is #)
automon = *3 ; One Touch Record a.k.a. Touch Monitor
parkcall = *4
;***multitenant callparking
#include /etc/asterisk/features.multiparking.conf

/etc/asterisk/features.multiparking.conf
[parkinglot_mhill]
context = mhillpark
parkpos = 1-9
findslot = first
parkinghints = yes ; Add hints priorities automatically for parking slots
(default is no).
parkedmusicclass = classical
parkingtime = 7200
parkedcalltransfers = both
parkedcallreparking = both

/etc/asterisk/extensions.conf
...
[parkinglot_mhill]
switch = Realtime/@extensions
...

/etc/asterisk/sip.conf
...
parkinglot=parkinglot_mhill
...
--
_
-- 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] atx timeout - play xferfailsound

2012-01-30 Thread John Taylor
Asterisk 1.6.2.20 on Debian Lenny

I'm finding that if no one answers an attended transfer (timeout set by
atxfernoanswertimeout), then the transferrer is handed back to the original
caller, and a beep is played.

In 1.4 I was able to indicate the timeout and failure by setting xferfailsound
to a custom recording, but this doesn't seem to happen in 1.6

How can I indicate a timeout to the transferrer?

Many thanks

John
--
_
-- 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] vigor 2920 problems

2012-01-30 Thread John Taylor
Thanks for help- suggestion fixed the issue

John


On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote:

 Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will
 get permission to try new firmware later!

 JT




 On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote:

 Hi John,

 We've had similiar issues with customers behind the 2920 connecting to a
 hosted asterisk system. If you rebooted a phone it often didn't
 re-register, Checking the NAT sessions table on the router revealed stale
 nat sessions open for the phone.

 On the advice of Dreytek we found a fix by lowering the NAT session
 timeout from the default of 24hrs down to 5 minutes and installing the
 latest release of the firmware (3.3.7) it may not be available on the UK
 Site at the moment (It wasn't when we did the upgrade!) but it can be got
 from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/

 It may help, It may not - But its quick easy fix if it does.

 Regards,
 AJ.


 - Original Message -
 From: John Taylor j...@vetsurgeon.org.uk
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, 21 November, 2011 10:20:14 AM
 Subject: [asterisk-users] vigor 2920 problems

 One of our clients has a Draytek Vigor 2920- their natted Snom phones
 behind it are registered to an Asterisk 1.4 server on an external public
 IP.

 I've set QOS, bandwidth management and turned off the SIP ALG via telnet
 but I'm still having some problems with some of the phones losing
 registration if Asterisk is restarted.

 I can see the phones sending SIP REGISTER messages, but they never
 arrive at the server; this happens in about half of the phones- with no
 consistency as to which lose registration.

 It looks like the router is swallowing the messages, or there's some
 kind of NAT problem. Other clients at other sites are fine.

 The problem clears if the phone is rebooted (renegotiates a new nat
 path?)

 Any help warmly appreciated.

 John

 --
 _
 -- 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] vigor 2920 problems

2011-11-21 Thread John Taylor
One of our clients has a Draytek Vigor 2920- their natted Snom phones
behind it are registered to an Asterisk 1.4 server on an external public IP.

I've set QOS, bandwidth management and turned off the SIP ALG via telnet
but I'm still having some problems with some of the phones losing
registration if Asterisk is restarted.

I can see the phones sending SIP REGISTER messages, but they never arrive
at the server; this happens in about half of the phones- with no
consistency as to which lose registration.

It looks like the router is swallowing the messages, or there's some kind
of NAT problem. Other clients at other sites are fine.

The problem clears if the phone is rebooted (renegotiates a new nat path?)

Any help warmly appreciated.

John
--
_
-- 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] vigor 2920 problems

2011-11-21 Thread John Taylor
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will
get permission to try new firmware later!

JT



On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote:

 Hi John,

 We've had similiar issues with customers behind the 2920 connecting to a
 hosted asterisk system. If you rebooted a phone it often didn't
 re-register, Checking the NAT sessions table on the router revealed stale
 nat sessions open for the phone.

 On the advice of Dreytek we found a fix by lowering the NAT session
 timeout from the default of 24hrs down to 5 minutes and installing the
 latest release of the firmware (3.3.7) it may not be available on the UK
 Site at the moment (It wasn't when we did the upgrade!) but it can be got
 from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/

 It may help, It may not - But its quick easy fix if it does.

 Regards,
 AJ.


 - Original Message -
 From: John Taylor j...@vetsurgeon.org.uk
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, 21 November, 2011 10:20:14 AM
 Subject: [asterisk-users] vigor 2920 problems

 One of our clients has a Draytek Vigor 2920- their natted Snom phones
 behind it are registered to an Asterisk 1.4 server on an external public
 IP.

 I've set QOS, bandwidth management and turned off the SIP ALG via telnet
 but I'm still having some problems with some of the phones losing
 registration if Asterisk is restarted.

 I can see the phones sending SIP REGISTER messages, but they never
 arrive at the server; this happens in about half of the phones- with no
 consistency as to which lose registration.

 It looks like the router is swallowing the messages, or there's some
 kind of NAT problem. Other clients at other sites are fine.

 The problem clears if the phone is rebooted (renegotiates a new nat
 path?)

 Any help warmly appreciated.

 John

 --
 _
 -- 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] intermittent problem on 1.4

2011-01-19 Thread John Taylor
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that
originated from a UK landline back up a SIP trunk to the same ITSP and on to
another UK landline number.

UK Landline-voipfone-asterisk 1.4-voipfone-UK landline

About 1 in 3 times the call at the final landline is silent and we see RTP
Read too short scrolling on the console log.

Where do we start working out what's going on? Other than that the server is
working well

John
--
_
-- 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] cannot answer incoming calls

2011-01-06 Thread John Taylor
Have recently installed some Snom phones into an office. Phones are
natted and connect to a 1.4 server on a public IP

We can make outgoing calls, but are unable to answer incoming calls.
The phone rings, but the call cannot be picked up. Other phones on
other sites connected to the server are working perfectly.

Looking at the SIP trace it appears the phone transmits:

Sent to udp:193.33.xx.xx:5060 at 6/1/2011 11:49:20:868 (849 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 193.33.xx.xx:5060;branch=z9hG4bK6e82052c;rport=5060
From: xx
sip:07765000...@sip3.office-voip.com;tag=as1b6fc27c
To: sip:x_...@79.123.xx.xx:25380;tag=37gg1zu3wp
Call-ID: 1b212085091e98387237125f0ab81...@sip3.office-voip.com
CSeq: 102 INVITE
Contact: sip:x_...@192.168.4.19:2048;reg-id=1
User-Agent: snom300/7.3.30
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 641540583 641540584 IN IP4 192.168.4.19
s=call
c=IN IP4 192.168.4.19
t=0 0
m=audio 52386 RTP/AVP 18 101
a=rtpmap:18 g729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

but it is never received by the server.

Interestingly RINGING and REGISTER messages are working OK. The NAT
router is out of our control. Are we looking at a SIP ALG getting in
the way?

Thanks,

John

--
_
-- 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] Record() Cmd and My SQL

2010-09-24 Thread John Taylor
Why not write the file to /tmp using MixMonitor, then use the command
option to trigger an AGI script that will move the data into your
database then delete the original file?

John

On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
 The reason is when doing a load balancing  , We  cannot confine the
 recording to a particular asterisk machine ( If we have more than one
 asterisk machine in the topology ).

 So a centralized mechanism might be better . So that any machine can
 access the recording .
 Regards
 Mahesh


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext David
 Backeberg
 Sent: Thursday, September 23, 2010 9:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Record() Cmd and My SQL

 On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
 mahesh.gov...@nsn.com wrote:
 HI ,

 Is there Any way is there so that I can store my recordings directly
 to a
 database rather storing the same to a file .

 Please, please, please tell us why you would want to do that.

 --
 _
 -- 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




-- 

JA Taylor
MA VetMB MRCVS
Mansion Hill Veterinary Practice
133-137 Main Road
Middleton Cheney OX17 2PP
01295 712110

-- 
_
-- 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] tcpdump auto stats script

2010-09-24 Thread John Taylor
Before I reinvent the wheel, I'm looking for a script then when run will
- launch tcpdump (or equivalent) on the server and capture all SIP and
UDP traffic to an IP address
- then, rather than me manually analysing with wireshark, will analyze
the cap file and produce stats on jitter, lag, delta etc.

Thanks for any help

John

-- 
_
-- 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] Snom phones recommended firmware

2010-09-04 Thread John Taylor
We're using firmware 7.3.30 on an installation of Snom 300 phones.
Should we stick with it, or do the newer firmwares have better support
for Asterisk?

Thanks

John

-- 
_
-- 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] Problem attended transfer with ilbc

2010-06-28 Thread John Taylor
I have an Asterisk server on our LAN that serves our office VOIP
phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are
ulaw/alaw

We use attended transfer extensively. If our trunk is ulaw/alaw they work fine.

If the trunk is ilbc we have problems
1- incoming PSTN call routed via voipfone SIP down the trunk to our server
2- our phones ring ok, caller can be answered (e.g. by A)
3- A requests attended transfer to another phone (B) on the LAN-
incoming caller put on hold, A can talk to B, B can talk to A
4- A hangs up, B is connected to caller. B can hear caller, but caller
cannot hear B. Console output:
Asked to transmit frame type 64, while native formats is 0x400
(ilbc)(1024) read/write = 0x40 (slin)(64)/0x400 (ilbc)(1024)

Running Asterisk 1.6.2.9 on Ubuntu Karmic- self compiled (do not seem
to be able to compile deb source package with ilbc, and deb package
does not have ilbc)

Any idea what may be happening?

John

-- 
_
-- 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] forward call back up same trunk to external cell phone problem

2010-02-01 Thread John Taylor
Hi- can anyone help with this. I'm really stuck as apparently it
should work. Is it a problem with the ITSP, with using the same trunk
for both legs of the call etc?

John

On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote:
 Hi

 If I have an incoming call coming down a SIP trunk to a particular
 internal SIP extension- I can answer the extension fine, all works
 well

 However, if I change extension.conf from dialling the internal
 extension to forward the call to an external cell phone (up the same
 trunk as the incoming leg of the call) I cannot get any audio and get
 the following error message on the console:
 [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short

 i.e. change from
 [voipfone_incoming]
 exten = s,1,Dial(SIP/203,20,t)

 to
 [voipfone_incoming]
 exten = s,1,Dial(SIP/07123123...@voipfone,20,t)

 What's wrong?!

 John


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] forward call back up same trunk to external cell phone problem

2010-01-30 Thread John Taylor
Hi

If I have an incoming call coming down a SIP trunk to a particular
internal SIP extension- I can answer the extension fine, all works
well

However, if I change extension.conf from dialling the internal
extension to forward the call to an external cell phone (up the same
trunk as the incoming leg of the call) I cannot get any audio and get
the following error message on the console:
[Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short

i.e. change from
[voipfone_incoming]
exten = s,1,Dial(SIP/203,20,t)

to
[voipfone_incoming]
exten = s,1,Dial(SIP/07123123...@voipfone,20,t)

What's wrong?!

John

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi,

I've now set dtmfmode=rfc2833 and that seems to have fixed it

John


2010/1/7 John Taylor j...@vetsurgeon.org.uk:
 We're now getting this problem on outgoing calls. I've forced the port
 to 100FD but still no joy. Anyone any ideas how to debug this- have
 added verbose to logger.conf

 Thanks for any help

 John

 2010/1/4 John Taylor j...@vetsurgeon.org.uk:
 I have recently moved our asterisk server from our LAN to a Debian
 Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
 network. Our phones are behind a natted firewall. An ITSP provides a
 PSTN to SIP termination for incoming calls

 Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x)

 Everything works fine (incoming/outgoing audio etc.) except
 occasionally an incoming caller is cut off whilst the called extension
 stays in the call and can hear a DTMF tone (multimon recognises it as
 tone D). The asterisk log file shows the call stays active despite
 the incoming caller being cut off. This has happened to all our
 extensions at some point (a combination of Snoms and Funkwerks). It
 happens fairly infrequently, and can happen at any point during a
 call.

 The public Lenny server's asterisk config is exactly the same as our
 LAN Ubuntu asterisk server where we never had this problem. The only
 difference is that the ITSP trunk is now ulaw rather than ilbc.

 Can anyone help? Relevant files below (trunk and extension codecs are both 
 ulaw)

 John


 example extension in sip.conf:
 [203]
 type=friend
 username=203
 secret=xx
 host=dynamic
 dtmfmode=inband
 call-limit=2
 qualify=yes
 nat=yes


 /var/log/asterisk/messages:
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00,
 08:30-18:00|mon-fri|*|*?day|100|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Goto (day,100,1)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Launched AGI Script
 /home/john/phpagi/lookup
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- AGI Script
 /home/john/phpagi/lookup completed, returning 0
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-09f74a00,
 CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:2] Monitor(SIP/301x-09f74a00,
 wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:4] Dial(SIP/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 203
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 206
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 207
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 220
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/206-0a005eb8 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/207-09fe2c98 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:59] VERBOSE[10712] logger.c:     -- SIP/203-0a001138
 answered SIP/301x-09f74a00



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com

Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi all,

I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it

John

2010/1/4 John Taylor j...@vetsurgeon.org.uk:
 I have recently moved our asterisk server from our LAN to a Debian
 Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
 network. Our phones are behind a natted firewall. An ITSP provides a
 PSTN to SIP termination for incoming calls

 Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x)

 Everything works fine (incoming/outgoing audio etc.) except
 occasionally an incoming caller is cut off whilst the called extension
 stays in the call and can hear a DTMF tone (multimon recognises it as
 tone D). The asterisk log file shows the call stays active despite
 the incoming caller being cut off. This has happened to all our
 extensions at some point (a combination of Snoms and Funkwerks). It
 happens fairly infrequently, and can happen at any point during a
 call.

 The public Lenny server's asterisk config is exactly the same as our
 LAN Ubuntu asterisk server where we never had this problem. The only
 difference is that the ITSP trunk is now ulaw rather than ilbc.

 Can anyone help? Relevant files below (trunk and extension codecs are both 
 ulaw)

 John


 example extension in sip.conf:
 [203]
 type=friend
 username=203
 secret=xx
 host=dynamic
 dtmfmode=inband
 call-limit=2
 qualify=yes
 nat=yes


 /var/log/asterisk/messages:
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00,
 08:30-18:00|mon-fri|*|*?day|100|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Goto (day,100,1)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Launched AGI Script
 /home/john/phpagi/lookup
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- AGI Script
 /home/john/phpagi/lookup completed, returning 0
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-09f74a00,
 CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:2] Monitor(SIP/301x-09f74a00,
 wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:4] Dial(SIP/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 203
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 206
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 207
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 220
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/206-0a005eb8 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/207-09fe2c98 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:59] VERBOSE[10712] logger.c:     -- SIP/203-0a001138
 answered SIP/301x-09f74a00


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] caller getting cut off intermittently

2010-01-07 Thread John Taylor
We're now getting this problem on outgoing calls. I've forced the port
to 100FD but still no joy. Anyone any ideas how to debug this- have
added verbose to logger.conf

Thanks for any help

John

2010/1/4 John Taylor j...@vetsurgeon.org.uk:
 I have recently moved our asterisk server from our LAN to a Debian
 Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
 network. Our phones are behind a natted firewall. An ITSP provides a
 PSTN to SIP termination for incoming calls

 Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x)

 Everything works fine (incoming/outgoing audio etc.) except
 occasionally an incoming caller is cut off whilst the called extension
 stays in the call and can hear a DTMF tone (multimon recognises it as
 tone D). The asterisk log file shows the call stays active despite
 the incoming caller being cut off. This has happened to all our
 extensions at some point (a combination of Snoms and Funkwerks). It
 happens fairly infrequently, and can happen at any point during a
 call.

 The public Lenny server's asterisk config is exactly the same as our
 LAN Ubuntu asterisk server where we never had this problem. The only
 difference is that the ITSP trunk is now ulaw rather than ilbc.

 Can anyone help? Relevant files below (trunk and extension codecs are both 
 ulaw)

 John


 example extension in sip.conf:
 [203]
 type=friend
 username=203
 secret=xx
 host=dynamic
 dtmfmode=inband
 call-limit=2
 qualify=yes
 nat=yes


 /var/log/asterisk/messages:
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00,
 08:30-18:00|mon-fri|*|*?day|100|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Goto (day,100,1)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Launched AGI Script
 /home/john/phpagi/lookup
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- AGI Script
 /home/john/phpagi/lookup completed, returning 0
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-09f74a00,
 CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:2] Monitor(SIP/301x-09f74a00,
 wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:4] Dial(SIP/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 203
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 206
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 207
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Called 220
 [Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
 of type 'SIP' (cause 3 - No route to destination)
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/206-0a005eb8 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/207-09fe2c98 is ringing
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:57] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/220-09fe7748 is ringing
 [Jan  4 09:58:58] VERBOSE[10712] logger.c:     -- SIP/203-0a001138 is ringing
 [Jan  4 09:58:59] VERBOSE[10712] logger.c:     -- SIP/203-0a001138
 answered SIP/301x-09f74a00


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] caller getting cut off intermittently

2010-01-04 Thread John Taylor
I have recently moved our asterisk server from our LAN to a Debian
Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
network. Our phones are behind a natted firewall. An ITSP provides a
PSTN to SIP termination for incoming calls

Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x)

Everything works fine (incoming/outgoing audio etc.) except
occasionally an incoming caller is cut off whilst the called extension
stays in the call and can hear a DTMF tone (multimon recognises it as
tone D). The asterisk log file shows the call stays active despite
the incoming caller being cut off. This has happened to all our
extensions at some point (a combination of Snoms and Funkwerks). It
happens fairly infrequently, and can happen at any point during a
call.

The public Lenny server's asterisk config is exactly the same as our
LAN Ubuntu asterisk server where we never had this problem. The only
difference is that the ITSP trunk is now ulaw rather than ilbc.

Can anyone help? Relevant files below (trunk and extension codecs are both ulaw)

John


example extension in sip.conf:
[203]
type=friend
username=203
secret=xx
host=dynamic
dtmfmode=inband
call-limit=2
qualify=yes
nat=yes


/var/log/asterisk/messages:
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new
stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
0?bankhols|200|1) in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00,
08:30-18:00|mon-fri|*|*?day|100|1) in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1)
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup)
in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script
/home/john/phpagi/lookup
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script
/home/john/phpagi/lookup completed, returning 0
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new
stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@macro-monitor:1] Set(SIP/301x-09f74a00,
CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352)
in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@macro-monitor:2] Monitor(SIP/301x-09f74a00,
wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m)
in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Executing
[...@day:4] Dial(SIP/301x-09f74a00,
SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Called 203
[Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
of type 'SIP' (cause 3 - No route to destination)
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Called 206
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Called 207
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- Called 220
[Jan  4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel
of type 'SIP' (cause 3 - No route to destination)
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing
[Jan  4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing
[Jan  4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing
[Jan  4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing
[Jan  4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing
[Jan  4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing
[Jan  4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138
answered SIP/301x-09f74a00

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] DNS reload on trunks for outgoing calls

2010-01-04 Thread John Taylor
Put the commonly used domain names + appropriate ips into /etc/hosts?

John

2010/1/4 Steve Howes steve-li...@geekinter.net:

 On 4 Jan 2010, at 08:34, Remco Barendse wrote:

 Is there any fix or workaround for the DNS problem (old standing bug
 that
 when the box starts and domain names do not resolve quickly enough
 from
 DNS then asterisk stops using the outgoing trunks.

 I read on the list before that it is considered a huge and
 unacceptable
 load for asterisk servers to try and resolve the domain names again
 after some time but it is rather annoying. I don't know about
 resources of other people but on my boxes i have some cpu cycles that
 could be used for that :)

 I now do nightly restarts of asterisk but it still means that at
 least for
 one day calls are flowing through expensive PSTN.

 If anybody knows of a workaround, would be most welcome

 Install a resolver locally.

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] multiple sip trunks

2009-12-15 Thread John Taylor
I thought so- the fact the server has 20 different registry entries to 20
different account all at the same ITSP shouldn't matter?

Can't see any DDI info in the SIP headers unfortunately :(

John

2009/12/14 meetmecall i...@meetmecall.nl

 The easiest solution to deal with this is to have one context with
 different extensions for the different numbers and route the incoming
 calls from there. It should look something like this (not a tested
 piece of asterisk script, just an example to give the idea).

 Hope it helps :-)


 Erik de Wild

 [all_trunks]

 exten = 31592123456,1,Goto(trunk1,s,1)
 exten = 31592123457,1,Goto(trunk1,s,1)
 exten = 31592123458,1,Goto(trunk1,s,1)

 exten = 3159212,1,Goto(trunk2,s,1)
 exten = 31592123334,1,Goto(trunk2,s,1)
 exten = 31592123335,1,Goto(trunk2,s,1)



 On 14 dec 2009, at 10:39, Olle E. Johansson wrote:

 
  11 dec 2009 kl. 23.21 skrev John Taylor:
 
  I have multiple trunks to the same ITSP. Incoming calls to any trunk
  go to the last incoming label defined in those trunks' contexts in
  sip.conf.
 
  My ITSP insists on insecure=very in the trunk context; is this the
  cause?
 
  This is an effect of the Asterisk architecture. We've had many
  discussions on how to change it, but right now the peer matching on
  IP/Port can't separate various instances from each other, since they
  all have the same IP/port. Asterisk simply goes for the first match,
  which happens to be the last entry with the IP/port in the sip.conf
  file.
 
  /Olle
  ___
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
Thanks - have done that and am now trying a one out. However, I'd
still like to know whether 1 asterisk server can support multiple
trunks/registry entries. Does it cause problems?

Thanks

John

2009/12/3 Tim Nelson tnel...@rockbochs.com:
 - John Taylor j...@vetsurgeon.org.uk wrote:
 I want to use an asterisk box to provide a voip service to a number
 of
 separate companies.

 I have a VOIP provider who I want to trunk with. As far as I can see
 it there are 2 options
 1. Have 1 SIP trunk to one account at the provider who gives me
 multiple incoming numbers; this is less than optimal as the provider
 does not provide the DID number in the sip header; I only get the
 account number. I have the option to set called line presentation
 but this will stop CLID

 2. Have multiple sip trunks to multiple accounts at the provider. Is
 this an advisable thing to do? I notice asterisk does not handle the
 incoming context correctly (all incoming calls go to the last
 incoming
 context defined in sip.conf), but I can extract the account called
 via
 the EXTEN variable.

 I would be looking at providing around 20 companies with accounts
 (all
 very small), and would prefer option (2) to enable failover to a
 number they specify.

 Thanks for any light shed

 John


 Why not go with a real carrier that can send you proper DID and DNIS 
 information for each call? Rather than trying to configure/code/etc around 
 the problem with the ITSP, use an ITSP that does things correctly. There are 
 many people here on asterisk-users that can recommend a proper ITSP. If you 
 want pure business response, head over to asterisk-biz and ask there.

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
I assume if all the SIP trunks are to the same host/port, Asterisk
cannot distinguish which trunk is active when an incoming call is
made- it will dump all incoming calls to the context specified in the
last trunk entry of sip.conf

Thanks

John

2009/12/11 Martin asteriskl...@callthem.info:
 On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote:
 Thanks - have done that and am now trying a one out. However, I'd
 still like to know whether 1 asterisk server can support multiple
 trunks/registry entries. Does it cause problems?
 yes, Asterisk does support multiple registry entries...
 if it didn't ... it would be just a crippled sip endpoint

 lets say more ... Asterisk can do whatever you want it to do (within
 reason and technical boundaries);
 just code it in or request a feature

 Martin


 Thanks

 John

 2009/12/3 Tim Nelson tnel...@rockbochs.com:
 - John Taylor j...@vetsurgeon.org.uk wrote:
 I want to use an asterisk box to provide a voip service to a number
 of
 separate companies.

 I have a VOIP provider who I want to trunk with. As far as I can see
 it there are 2 options
 1. Have 1 SIP trunk to one account at the provider who gives me
 multiple incoming numbers; this is less than optimal as the provider
 does not provide the DID number in the sip header; I only get the
 account number. I have the option to set called line presentation
 but this will stop CLID

 2. Have multiple sip trunks to multiple accounts at the provider. Is
 this an advisable thing to do? I notice asterisk does not handle the
 incoming context correctly (all incoming calls go to the last
 incoming
 context defined in sip.conf), but I can extract the account called
 via
 the EXTEN variable.

 I would be looking at providing around 20 companies with accounts
 (all
 very small), and would prefer option (2) to enable failover to a
 number they specify.

 Thanks for any light shed

 John


 Why not go with a real carrier that can send you proper DID and DNIS 
 information for each call? Rather than trying to configure/code/etc around 
 the problem with the ITSP, use an ITSP that does things correctly. There 
 are many people here on asterisk-users that can recommend a proper ITSP. If 
 you want pure business response, head over to asterisk-biz and ask there.

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.

My ITSP insists on insecure=very in the trunk context; is this the cause?

John

2009/12/11 Noah Miller noahisaacmil...@gmail.com:
 I assume if all the SIP trunks are to the same host/port, Asterisk
 cannot distinguish which trunk is active when an incoming call is
 made- it will dump all incoming calls to the context specified in the
 last trunk entry of sip.conf

 No.  SIP uses authentication (well, I guess you can not use
 authentication).  Asterisk (and almost any SIP gateway) will correctly
 match the call to the trunk based on the authentication.  Even if you
 didn't send any authentication info, asterisk will try to match the
 call as a guest call.  It is common practice to not allow
 unauthenticated SIP traffic.


 - Noah

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] multiple sip trunks

2009-12-03 Thread John Taylor
I want to use an asterisk box to provide a voip service to a number of
separate companies.

I have a VOIP provider who I want to trunk with. As far as I can see
it there are 2 options
1. Have 1 SIP trunk to one account at the provider who gives me
multiple incoming numbers; this is less than optimal as the provider
does not provide the DID number in the sip header; I only get the
account number. I have the option to set called line presentation
but this will stop CLID

2. Have multiple sip trunks to multiple accounts at the provider. Is
this an advisable thing to do? I notice asterisk does not handle the
incoming context correctly (all incoming calls go to the last incoming
context defined in sip.conf), but I can extract the account called via
the EXTEN variable.

I would be looking at providing around 20 companies with accounts (all
very small), and would prefer option (2) to enable failover to a
number they specify.

Thanks for any light shed

John

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] trunk peer not registering after migrating installation

2008-11-27 Thread John Taylor
I have an odd problem. I have just installed asterisk on an ubuntu
box, and migrated the previous configuration of asterisk (on another
ubuntu box) to this new server (scp -pr [EMAIL PROTECTED]:/etc/asterisk/*
/etc/asterisk/)

Asterisk worked fine on the old server, but on this server my SIP
trunk peer does not login after initial server startup. sip show
peers shows  my phones registered OK, but the peer describing my SIP
trunk does not even display:
sip show peers
Name/username  HostDyn Nat ACL Port Status
204/204192.168.xxx.xxx   D  2048 Unmonitored
203/203192.168.xxx.xxx   D  2048 Unmonitored
sip show registry
sip.voipfone.co.uk:5060 45 Registered
 Thu, 27 Nov 2008 11:01:56:03

sip reload or restarting asterisk with /etc/init.d/asterisk restart
fixes the problem and I get the following output:
Name/username  HostDyn Nat ACL Port Status
204/204192.168.xxx.xxx   D  2048 Unmonitored
203/203192.168.xxx.xxx  D  2048 Unmonitored
voipfone/  195.189.173.10  5060 OK (61 ms)

sip show registry
sip.voipfone.co.uk:5060 45 Registered
 Thu, 27 Nov 2008 11:05:28:02

sip.conf entry for the trunk
[voipfone]
type=friend
secret=xx
username=
fromuser=
fromdomain=sip.voipfone.co.uk
host=sip.voipfone.co.uk
insecure=very
dtmfmode=rfc2833
context=fromvoipfone ;inbound calls falls in this context of dialplan
disallow=all
allow=ilbc
;allow=ulaw
;allow=alaw
qualify=yes

Any ideas warmly welcomed! Setting debug to level 9 isn't helping me
out on this.

John

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] call queuing not detecting caller hang up when call originates from voip provider

2007-12-28 Thread John Taylor
Dear all

I've got call queuing working when calls originate from my local site.

After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a phantom until one of the extensions answers it and
it is destroyed

Am I doing something wrong? Am using asterisk 1.4.16.2

Relevant part of files:

sip.conf

[voipfone]
type=friend
secret=
username=xx
fromuser=xx
fromdomain=sip.voipfone.co.uk
host=sip.voipfone.co.uk
insecure=very
dtmfmode=rfc2833
context=fromvoipfone

[s450]
type=friend
context=phones
host=dynamic

[xlite]
type=friend
context=phones
host=dynamic

[consult]
type=friend
context=phones
host=dynamic

extensions.conf

[fromvoipfone]
exten= 1234,1,Dial(SIP/consult,3)
exten= 1234,n,Answer
exten= 1234,n,Ringing
exten= 1234,n,Wait(2)
exten= 1234,n,Background(/var/lib/asterisk/sounds/mhqw)
exten= 1234,n,Queue(myqueue|r)
exten= 1234,n,Hangup

[phones]
exten= 1001,1,Dial(SIP/s450)
exten= 1002,1,Dial(SIP/xlite)
exten= 1003,1,Dial(SIP/consult)
exten= _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,r)
exten= _ZX,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)
exten= _Z,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)

queues.conf

[myqueue]
periodic-announce = mhqw
periodic-announce-frequency = 10
music=default
strategy=ringall
timeout=15
retry=5
wrapuptime=0
maxlen=0
announce-frequency=0
announce-holdtime=no
member = SIP/consult,1
context = phones

Any help appreciated!!

John

___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users