from:"Gavin Henry"

[asterisk-users] Commercial SSL certs on Asterisk 1.8.10.0 with Polycom phones for encrypted calls using TLS and SRTP?

2012-03-08 Thread Gavin Henry

Hi all,

We're testing TLS and SRTP on Asterisk 1.8.10.0 and have it working
with a commerical (not self-sign) AlphaSSL wildcard (GlobalSign) using
Blink Lite 1.6.2 as per
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial

We've tested with Bria on an iPhone and that doesn't recognised the
commercial CA (GlobalSign Root CA).

On a Yealink 28P with V60/V61 is registers over TLS, but can't do
SRTP. Yealink are working on this and are testing against one of our
dev servers.

My question is someone (Digium) must have this working against Polycom
(which is a requirement for this project) with commercial certs since
that's their partner of choice?

This is our relevant setup:

tlsenable=yes
tlsbindaddr=0.0.0.0
tcpbindaddr=0.0.0.0
tcpenable=yes
transport=tcp,udp,tls
tlscertfile=/etc/asterisk/ssl/test_wildcard_cert.pem
tlscafile=/etc/asterisk/ssl/AlphaSSLroot.crt
tlscipher=ALL
tlsclientmethod=tlsv1


This file has the cert and key in it:

test_wildcard_cert.pem

is as per:

http://www.alphassl.com/support/install-ssl/apache.html

and AlphaSSLroot.crt is as per:

http://www.alphassl.com/support/install-root/apache.html

We haven't tested Snom or Aastra yet.

Thanks,

Gavin.


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Re: [asterisk-users] Commercial SSL certs on Asterisk 1.8.10.0 with Polycom phones for encrypted calls using TLS and SRTP?

2012-03-08 Thread Gavin Henry


 My question is someone (Digium) must have this working against Polycom
 (which is a requirement for this project) with commercial certs since
 that's their partner of choice?


 I don't believe we've done any interop testing with Polycom phones since TLS
 and SRTP support were added to Asterisk. Most (possibly all) of the interop
 testing was done with Asterisk Business Edition, the last version of which
 was based on Asterisk 1.4.


Ah, this makes sense now. So as of today the status of TLS and SRTP in anything
other than 1.4.X is unknown?

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Re: [asterisk-users] Commercial SSL certs on Asterisk 1.8.10.0 with Polycom phones for encrypted calls using TLS and SRTP?

2012-03-08 Thread Gavin Henry


 Ah, this makes sense now. So as of today the status of TLS and SRTP in
 anything
 other than 1.4.X is unknown?


 Umm... no :-)

OK, sorry :-)

 Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of
 these were tested with Polycom phones the last time we did interop testing
 with those phones.

Ah, I forgot when it was added.

 The status of SIP/TLS and SRTP support in the Asterisk releases that have
 them are not 'unknown'; they are there and expected to be working. I was
 just pointing out that Digium has not specifically tested Polycom phones for
 interop with these features, and certainly has not specifically tested usage
 of TLS certificates issued by any particular CA.

Has anyone on the list?

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Re: [asterisk-users] SipSak: Send SIP OPTION with password

2010-10-23 Thread Gavin Henry

It's replying so its up :)
On 23 Oct 2010 17:32, Jonas Kellens jonas.kell...@telenet.be wrote:
 Hello,

 I'm trying to use SipSak to check if my Asterisk server is
 available/running with the following :

 sipsak -vv -s sip:usern...@sip.domain.tld -c sip:usern...@sip.domain.tld
 --password guessthis --hostname XX.XX.XX.63

 The SIP OPTION is received by Asterisk as follow :

 OPTIONS sip:usern...@sip.domain.tld SIP/2.0
 Via: SIP/2.0/UDP XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;rport;alias
 *From: sip:sip...@xx.xx.xx.63:36887;tag=5e8faf01*
 To: sip:usern...@sip.domain.tld
 Call-ID: 1586474...@xx.xx.xx.63
 CSeq: 1 OPTIONS
 Contact: sip:sip...@xx.xx.xx.63:36887
 Content-Length: 0
 Max-Forwards: 70
 User-Agent: sipsak 0.9.6
 Accept: text/plain


 and it send back :

 SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP

XX.XX.XX.63:36887;branch=z9hG4bK.304f1a46;alias;received=XX.XX.XX.63;rport=36887
 *From: sip:sip...@xx.xx.xx.63:*36887*;tag=5e8faf01*
 To: sip:usern...@sip.domain.tld;tag=as29357d12
 Call-ID: 1586474...@xx.xx.xx.63
 CSeq: 1 OPTIONS
 Server: Asterisk PBX 1.6.2.10
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Accept: application/sdp
 Content-Length: 0


 I am not able to change the FROM-header so Asterisk authenticates the
 OPTION being sent.


 Jonas.

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[asterisk-users] Commands needed via AMI to find callerid of inbound call to extension

2010-09-20 Thread Gavin Henry

Hi all,

Can anyone help with the logic of which commands to use to say:

1. Extension is 600
2. See if has an ongoing call
3. Check if inbound or outbound to the extension
4. Find callerid of inbound call

Been reading http://www.voip-info.org/wiki/view/Asterisk+manager+API

Using latest 1.6.

Thanks.

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Re: [asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions

2010-04-24 Thread Gavin Henry

Hi,

I look after this but have been very busy for months. Maybe you canhelp me test?

Thanks,

Gavin.

On 23/04/2010, Sean Brady sbr...@gtfservices.com wrote:
 Not sure if this is the right place to ask, but what do we need to do to
 get this patch merged?  How can I help?  I'm no dev, but I use LDAP with
 Asterisk and I might be of some help.

 Thanks guys.

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Re: [asterisk-users] BT ISDN-30 Call Failures

2010-03-25 Thread Gavin Henry

Any probs with the circuits?

Try and upgrade?

On 17/03/2010, Russell Brown russ...@lls.lls.com wrote:


 I'm seeing both inbound and outgoing call failures on our ISDN-30 lines
 that only seem to go away when I do a zap restart or in extremis
 restart Asterisk (1.4.25 with a Digium TE205P and zaptel 1.4.12.1).  If
 I don't restart zapata or Asterisk the problem rapidly get worse :-(

 The lines are from BT with LCR from CableWireless (I've tried using the
 LCR bypass code and it doesn't make any difference).

 The outbound symptoms are that the number appears to go out on the ISDN
 but the caller hears nothing and the callee's phone doesn't ring
 (example below with ISDN debug on).

 The incoming problem is that the callee's phone rings for a couple of
 seconds and then the call gets cutoff (again example enclosed) or
 sometimes doesn't appear at all in the Asterisk logs and the callee gets
 a busy tone (no the system hasn't used all of the channels at this point
 :-).

 Can anyone suggest a cause and/or remedy?  Any idea what the
 disconnection stuff in the PRI debug means?



 Outbound log extract:


  [Mar 17 16:58:02] VERBOSE[6630] logger.c: -- Executing
 [01780471...@from-sip:18] Dial(SIP/197-b6726980,
 Zap/G1/01780471800||TWK) in new stack
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: -- Making new call for cr 33089
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: -- Requested transfer
 capability: 0x00 - SPEECH
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Protocol Discriminator: Q.931
 (8)  len=40
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Call Ref: len= 2 (reference
 321/0x141) (Originator)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Message type: SETUP (5)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  [04 03 80 90 a3]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Bearer Capability (len= 5) [
 Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: 
 Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: 
 User information layer 1: A-Law (35)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  [18 03 a9 83 9e]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Channel ID (len= 5) [ Ext: 1
 IntID: Implicit  PRI  Spare: 0  Exclusive  Dchan: 0
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: ChanSel:
 As indicated in following octets
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:Ext: 1
 Coding: 0  Number Specified  Channel Type: 3
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:Ext: 1
 Channel: 30 ]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  [6c 08 00 80 38 34 36 30 38 30]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Calling Number (len=10) [ Ext:
 0  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: 
 Presentation: Presentation permitted, user number not screened (0)  '846080'
 ]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  [70 0c 80 30 31 37 38 30 34 37
 31 38 30 32]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Called Number (len=14) [ Ext: 1
  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '01780471800' ]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  [a1]
  [Mar 17 16:58:02] VERBOSE[6630] logger.c:  Sending Complete (len= 1)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: q931.c:3134 q931_setup: call
 33089 on channel 30 enters state 1 (Call Initiated)
  [Mar 17 16:58:02] VERBOSE[6630] logger.c: -- Called G1/01780471800
 *** At this point the caller hears nothing and the phone on
   *** 01780471800 *doesn't* receive a call.
  [Mar 17 16:58:50] VERBOSE[6630] logger.c: NEW_HANGUP DEBUG: Calling
 q931_hangup, ourstate Call Initiated, peerstate Overlap sending
  [Mar 17 16:58:50] VERBOSE[6630] logger.c: q931.c:3015 q931_disconnect: call
 33089 on channel 30 enters state 11 (Disconnect Request)
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:  Protocol Discriminator: Q.931
 (8)  len=9
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:  Call Ref: len= 2 (reference
 321/0x141) (Originator)
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:  Message type: DISCONNECT (69)
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:  [08 02 81 90]
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:  Cause (len= 4) [ Ext: 1
 Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network
 serving the local user (1)
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:   Ext: 1  Cause:
 Normal Clearing (16), class = Normal Event (1) ]
  [Mar 17 16:58:50] VERBOSE[6630] logger.c: -- Hungup 'Zap/30-1'
  [Mar 17 16:58:50] VERBOSE[6630] logger.c:   == Spawn extension (from-sip,
 01780471800, 18) exited non-zero on 'SIP/197-b6726980'


 Here's an example of the inbound failure:


  [Mar 17 17:04:46] VERBOSE[13006] logger.c: -- Executing
 [846...@isdn_in:1] Ringing(Zap/20-1, ) in new stack
  [Mar 17 17:04:46] VERBOSE[13006] logger.c: q931.c:2844 q931_alerting: call
 79 on channel 20 enters state 7 (Call Received)
  [Mar 17

Re: [asterisk-users] Better SIP security please! Was: (no subject)

2010-03-21 Thread Gavin Henry

Has anyone done this with OpenSIPS? For example where it fronts an
Asterisk cluster with the load balancer module?

Thanks,

Gavin.


On 19/03/2010, Ryan Bullock rrb3...@gmail.com wrote:

 Hey Philipp,


 You can check out
 http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk for
 setting up from brute force detection and blocking with asterisk. There are
 also a link at the bottom about rate limiting registrations via iptables.


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Re: [asterisk-users] A2Billing and other prepaid Billing like ASTCC, who is better?

2010-02-06 Thread Gavin Henry

Why not pay for missing feature and contribute them to the project.
It's a very good product.


On 06/02/2010, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 I used A2Billing, basically it is nice and fine, but management
 possibilities is not that rich, so a lot of staff are need to be repeated
 that let the admin facing a problem of the needed time to do the task.

 Anyone advise for another open source prepaid billing that is rich by the
 management features?

 Also, I hope to find an open source Billing (prepaid and postpaid) that can
 work with Asterisk and Gnugk at the same time (instead of using one billing
 for asterisk and one billing for gnugk, specially that gnugk is good for
 h323 functionalities that are missing in asterisk).

 Appreciate any help and advise in that direction.

 Regards
 Bilal




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Re: [asterisk-users] Realtime LDAP Queues crashes

2010-01-07 Thread Gavin Henry

What are the LDAP searches like?

On 05/01/2010, Jorge Salamero Sanz ben...@cauterized.net wrote:
 Hi all,

 I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
 attributes needed for a working LDAP backend (I'll open a bug to include
 these
 changes on svn).

 SIP users and dialplan are perfectly working, but when I call a queue the
 whole Asterisk (1.6.2.0) crashes:

 on extconfig:

 [settings]
 sipusers = ldap,dc=nodomain,sip
 sippeers = ldap,dc=nodomain,sip
 extensions = ldap,dc=nodomain,extensions
 voicemail = ldap,dc=nodomain,voicemail
 queue_members = ldap,dc=nodomain,queue_member
 queues = ldap,dc=nodomain,queue

 on res_ldap.conf: see [1]

 for the Queues on LDAP I have:

 ou=Queues,dc=nodomain
 ou: Queues
 objectClass: top
 objectClass: organizationalUnit

 cn=foobar,ou=Queues,dc=nodomain
 objectClass: applicationProcess
 objectClass: AsteriskQueue
 AstQueueName: foobar
 AstQueueContext: default
 AstQueueTimeout: 180
 cn: foobar

 the dialplan (on extensions.conf, the same if it's on LDAP):

 [frontdesk]
 exten = 78,1,Answer
 exten = 78,n,Queue(foobar)
 exten = 78,n,Hangup

 [default]
 include = common
 include = frontdesk
 switch = Realtime

 and the user on LDAP:

 uid=foo,ou=Users,dc=nodomain
 cn: foo foo
 uid: foo
 sn: foo
 uidNumber: 2002
 gidNumber: 1901
 homeDirectory: /nonexistent
 userPassword: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
 eboxSha1Password: {SHA}C+7Hteo/D9vJXQ3UfzxbwnXaijM=
 eboxMd5Password: {MD5}rL0Y20zC+Fzt72VPzMSk2A==
 eboxLmPassword: 5BFAFBEBFB6A0942AAD3B435B51404EE
 eboxNtPassword: AC8E657F83DF82BEEA5D43BDAF7800CC
 eboxDigestPassword: {MD5}x0Z+Prb70OIF3iARsuJ3Xg==
 eboxRealmPassword: {MD5}c7467e3eb6fbd0e205de2011b2e2775e
 givenName: foo
 description: foo
 AstAccountType: friend
 AstAccountContext: users
 AstAccountCallerID: 1001
 AstAccountMailbox: 1001
 AstAccountHost: dynamic
 AstAccountNAT: yes
 AstAccountQualify: yes
 AstAccountCanReinvite: no
 AstAccountDTMFMode: rfc2833
 AstAccountInsecure: port
 AstAccountLastQualifyMilliseconds: 0
 AstAccountIPAddress: 0.0.0.0
 AstAccountPort: 0
 AstAccountExpirationTimestamp: 0
 AstAccountRegistrationServer: 0
 AstAccountUserAgent: 0
 AstAccountFullContact: sip:0.0.0.0
 AstContext: users
 AstVoicemailMailbox: 1001
 AstVoicemailPassword: 1001
 AstVoicemailEmail: u...@domain
 AstVoicemailAttach: yes
 AstVoicemailDelete: no
 AstQueueMembername: foobar
 AstQueueMemberof: foobar
 objectClass: AsteriskQueueMember
 objectClass: AsteriskSIPUser
 objectClass: AsteriskVoiceMail
 objectClass: inetOrgPerson
 objectClass: passwordHolder
 objectClass: posixAccount
 AstQueueInterface: SIP/1001

 when i call the queue extension, on slapd I can see how Asterisk fetches the
 AsteriskQueue objectClass, and then fetches the foo user, but then crashes
 like this:

 -- Executing [...@users:1] Answer(SIP/demo-, ) in new stack
 -- Executing [...@users:2] Queue(SIP/demo-, foobar) in new
 stack
 [Jan  5 13:26:28] WARNING[6195]: app_queue.c:1134 create_queue_member: No
 location at interface ''
 [1]6124 segmentation fault (core dumped)  asterisk -
 vvc

 *CLI queue show foobar
 [1]6356 segmentation fault (core dumped)  asterisk -
 vvc

 *CLI queue add member SIP/foo to foobar
 [1]6394 segmentation fault (core dumped)  asterisk -
 vvc

 any clue on what's wrong ? how could i debug this ? maybe there is some
 attribute missing ? or the LDAP schema is wrong ? anyone with a working
 setup
 like this ?

 thanks in advance !

 [0] http://people.ebox-platform.com/~bencer/asterisk.ldif
 [1] http://people.ebox-platform.com/~bencer/res_ldap.conf.mas

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Re: [asterisk-users] LDAP integration

2009-09-29 Thread Gavin Henry

Which version of the LDAP schema? I look after the one in 1.6.

Thanks.

On 29/09/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
 On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote:
 Hi all,

 I looked on the Internet but I didn't find any good how-to.
 I would like to integrate a ldap server ( with all users data) with
 asterisk to authenticate SIP users. With this solution I will only
 need to add a user on ldap, it will not be necessary to add any
 special configuration on sip.conf

 Is that possible???If so, How can I configure this setup???

 Thanks in advance

 I considered doing this using LDAP as a real-time database.  I decided
 not to for two reasons which I'll share below. However, I am very new to
 Asterisk so I would be very curious to know from more experienced folks
 if my assumptions were false.

 First, there were some good how-tos about using LDAP as a real-time
 database but, if I recall, the schema is extended in such a way that the
 regular user password is not the password used by Asterisk.

 Second, I believe we saw a way we could map the Asterisk password to the
 regular user password (it's been a while so I'm not sure about that) but
 were concerned about the problems of entering secure passwords from a
 phone keypad.  We enforce fairly secure passwords - at least nine
 characters with some variety of characters and encourage much longer
 passwords.  Having to enter lots of characters in both cases as well as
 symbols seemed difficult from a phone keypad.  Thus, we decided
 (reluctantly) to use separate simple passwords for phone access instead
 of the very secure passwords we use to data access.

 Hope this helps and looking forward to more informed comments than mine!
 - John
 --
 John A. Sullivan III
 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com

 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society


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Re: [asterisk-users] dCAP Exam

2009-09-20 Thread Gavin Henry

Aastra phones need reboots too :-(

On 20/09/2009, Alex Balashov abalas...@evaristesys.com wrote:
 Philipp Kempgen wrote:

 IMHO the Polycoms are a bad choice for the test because they
 reboot for every modification of the SIP account parameters so
 unless you have previous experience with the Polycoms you will
 loose a lot of time.

 Yeah, tell me about it.

 Snom is where it's at for instant provisioning changes.

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 Tel : (+1) (678) 954-0670
 Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] LDAP Get for Asterisk 1.6.x

2009-08-26 Thread Gavin Henry

2009/8/24 David Klaverstyn d...@klaverstyn.com.au:
 I’d appreciate it if someone could give me an answer to using LDAP in
 Asterisk 1.6.x

You can use res_config_ldap for storing Asterisk data in a directory
server for the realtime framework.

Thanks.


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[asterisk-users] ntop and Asterisk

2009-08-06 Thread Gavin Henry

Hi,

Would it be sane to run ntop on the same box as Asterisk or better to
mirror a LAN port etc?

http://www.ntop.org/OpenSourceVoipMonitoring.pdf

Thanks.

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[asterisk-users] BT IP Exchange interconnect

2009-07-31 Thread Gavin Henry

Hi All,

Has anyone passed the tests using Asterisk:

http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html

I presume the same rules apply for scaling and possibly have
OpenSIPS/Kamailio on the front?

Thanks.

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Re: [asterisk-users] BT IP Exchange interconnect

2009-07-31 Thread Gavin Henry

2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
 On Fri, 31 Jul 2009, Gavin Henry wrote:

 Hi All,

 Has anyone passed the tests using Asterisk:

 http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html

 Intersting. Looks like BT trying to become an ITSP to compete with the
 other ITSPs in the UK who already have PSTN wholesale interconnect...

 And they've done it in typical BT corprat style too - document upon
 document (30 pages for the credit check one!)  Extend, Embrace,
 Extinguish...

Yeah, true. I know some on the FreeSWITCH lists have passed all tests.

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Re: [asterisk-users] BT IP Exchange interconnect

2009-07-31 Thread Gavin Henry

2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
 On Fri, 31 Jul 2009, Gavin Henry wrote:

 Hi All,

 Has anyone passed the tests using Asterisk:

 http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html

 Intersting. Looks like BT trying to become an ITSP to compete with the
 other ITSPs in the UK who already have PSTN wholesale interconnect...

 And they've done it in typical BT corprat style too - document upon
 document (30 pages for the credit check one!)  Extend, Embrace,
 Extinguish...

What worries me is which way to go? According to Ofcom today:

http://www.ofcom.org.uk/consult/condocs/ngndevelopments/summary/

BT are going to be running IPX (eXchange) along with current TDM to
PSTN interconnects for a long time until their kit reaches endoflife.

So I see you getting the same as you would terminating with a BT
wholesale customer with TDM to IP as going direct to BT IP Exchange,
but not obviously getting the same minute price. You would save all
the legal process (and 9 months) going with an existing termination
provider though.

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Re: [asterisk-users] BT IP Exchange interconnect

2009-07-31 Thread Gavin Henry

2009/7/31 Steve Howes st...@geekinter.net:

 On 31 Jul 2009, at 08:22, Gavin Henry wrote:
 Has anyone passed the tests using Asterisk:

 BT guy we spoke to said yes : )

Good to know!


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Re: [asterisk-users] DAHDI - analogue, not seeing ringing (UK)

2009-07-31 Thread Gavin Henry

2009/7/31 Gordon Henderson gordon+aster...@drogon.net:
 On Fri, 31 Jul 2009, --[ UxBoD ]-- wrote:

 Gordon,

 Cast your mind back as I had a similar issue ... changing the cable sorted 
 it for me!

 Cursiously enough, I thought about that - but these were 2 brand new
 cables out of packets and I did check to see that they only had 2 wires
 connected at both ends. But I'll get the chaps on-site to check it again
 and see what they say.

 Still at a loss as to what a 4-wires cable does. Are there more than 2
 pins connected at the TDM card end?

What does your dahdi config look like?


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Re: [asterisk-users] Truecall

2009-07-18 Thread Gavin Henry

Exactly. I was thinking that a similar service would be a good addon
as an option to an ITSP.

Gavin.

On 18/07/2009, Steve Totaro stot...@totarotechnologies.com wrote:
 On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News)
 alansli...@gmail.comwrote:

 On 18/07/09 00:35, Gavin Henry wrote:
  This has to be an Asterisk based appliance no?
 
  http://www.truecall.co.uk/acatalog/trueCall_Features.html

 I saw this on the TV the other night. Couldn't believe how the dragons
 all thought it was such a cool idea.

 I was shouting at the telly saying You could do that with Asterisk very
 easily...

 Granted, if he's made the box, built it on an embedded SoC device then
 fair play, but he needs to have something Unique or anyone can do it.

 Alan


 Neat little box.

 In today's world, anyone can do just about anything.

 Anything unique doesn't stay that way for very long as soon as someone
 else takes notice of a a new unique thingy that is profitable.

 It is all about packaging and marketing.

 There is plenty of space in most markets for dozens of competitors, and it
 actually a great thing.

 --
 Thanks,
 Steve Totaro
 +18887771888 (Toll Free)
 +12409381212 (Cell)
 +12024369784 (Skype)


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Re: [asterisk-users] Truecall

2009-07-18 Thread Gavin Henry

Yeah, and the fxs port too.

On 18/07/2009, Alan Lord (News) alansli...@gmail.com wrote:
 On 18/07/09 00:35, Gavin Henry wrote:
 This has to be an Asterisk based appliance no?

 http://www.truecall.co.uk/acatalog/trueCall_Features.html

 I saw this on the TV the other night. Couldn't believe how the dragons
 all thought it was such a cool idea.

 I was shouting at the telly saying You could do that with Asterisk very
 easily...

 Granted, if he's made the box, built it on an embedded SoC device then
 fair play, but he needs to have something Unique or anyone can do it.

 Alan


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[asterisk-users] Truecall

2009-07-17 Thread Gavin Henry

This has to be an Asterisk based appliance no?

http://www.truecall.co.uk/acatalog/trueCall_Features.html

Looks pretty easy to setup using AstLinux or similar.

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Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

2009-06-07 Thread Gavin Henry

That is correct. That is the first test we did.

On 07/06/2009, Moises Silva moises.si...@gmail.com wrote:
 On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote:
 Every call as soon as the sangoma card is live.

 Speak to Konrad on your techdesk for more info.

 Thanks.


 I'll speak with him on Monday.

 However if you can provide more information before Monday I will be
 able to think beforehand on this matter.

 So please confirm this. If you get an incoming call and send it to
 Playback(demo-congrats) and then receive a second call and send it to
 Playback(tt-monkeys), both callers will listen both demo-congrats and
 tt-monkeys sounds?

 --
 Moises Silva
 Software Developer
 Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
 L3R 9T3 Canada
 t. 1 905 474 1990 x 128 | e. m...@sangoma.com

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[asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

2009-06-06 Thread Gavin Henry

Hi,

Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?

We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.

We are about to try the card and four *seperate* UK BT lines in a 32bit system.

The current system is a 4gb, dual core cpu with  pbx in a flash 1.4,
Zaptel and Asterisk 1.4.21-2

Currently we have put in a temp OpenVOX tdm400 card and it works
perfectly. As soon as we swap that and use Sangoma via wanrouter we
get crosstalk. For example, if an existing call is happening and a new
internal to external call or vise versa happens, they can hear each
other, even just to IVR.

Any ideas? All wiring has been checked and this *does not*, I repeat,
*does not* happen with the Sangoma card. So what ever explaination we
come up with, that fact remains and we get stumped.

Oh, the card and four fxo modules have been completely replaced and
64bit has been compiled in the wanrouter driver and Sangoma tech
support have ran out of suggestions. We have also tried going down to
2gb on the 64bit system too.

Hopefully 32bit will work, but we have other clients on 64bit with
Sangoma and they work. What is the Sangoma latest stable 64bit driver
doing!

Thanks,

Gavin.

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[asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

2009-06-06 Thread Gavin Henry

Hi,

Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?

We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.

We are about to try the card and four *seperate* UK BT lines in a 32bit system.

The current system is a 4gb, dual core cpu with  pbx in a flash 1.4,
Zaptel and Asterisk 1.4.21-2

Currently we have put in a temp OpenVOX tdm400 card and it works
perfectly. As soon as we swap that and use Sangoma via wanrouter we
get crosstalk. For example, if an existing call is happening and a new
internal to external call or vise versa happens, they can hear each
other, even just to IVR.

Any ideas? All wiring has been checked and this *does not*, I repeat,
*does not* happen with the Sangoma card. So what ever explaination we
come up with, that fact remains and we get stumped.

Oh, the card and four fxo modules have been completely replaced and
64bit has been compiled in the wanrouter driver and Sangoma tech
support have ran out of suggestions. We have also tried going down to
2gb on the 64bit system too.

Hopefully 32bit will work, but we have other clients on 64bit with
Sangoma and they work. What is the Sangoma latest stable 64bit driver
doing!

Thanks,

Gavin.

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Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

2009-06-06 Thread Gavin Henry

Every call as soon as the sangoma card is live.

Speak to Konrad on your techdesk for more info.

Thanks.

On 06/06/2009, Moises Silva moises.si...@gmail.com wrote:
 Currently we have put in a temp OpenVOX tdm400 card and it works
 perfectly. As soon as we swap that and use Sangoma via wanrouter we
 get crosstalk. For example, if an existing call is happening and a new
 internal to external call or vise versa happens, they can hear each
 other, even just to IVR.
 How often does this happen? (the cross-talk) every single call? is
 easy to reproduce?


 Any ideas? All wiring has been checked and this *does not*, I repeat,
 *does not* happen with the Sangoma card. So what ever explaination we
 come up with, that fact remains and we get stumped.
 You meant that this does not happen with the OpenVox card, didn't you?
 otherwise, you lost me.

 If you can easily reproduce this, I'd be interested in look into it.

 --
 Moises Silva
 Software Developer
 Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
 L3R 9T3 Canada
 t. 1 905 474 1990 x 128 | e. m...@sangoma.com

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

Where do they currently change their password? If it's somewhere you
control, why not add some to create the realmed password?

Gavin.

On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hello, all. I'm afraid I've been dropped into the deep end even though
I am an Asterisk novice. I've set up a few tiny, tiny systems in the
past and have now been asked to pull together Asterisk, FreePBX,
Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service.

After googling and reading for most of the last 24 hours, I finally have
my head around the components and how they work but am a little stumped
by password synchronization using existing LDAP accounts. Maintaining
separate accounts with a shared database between Kamailio and Asterisk
seems quite reasonable. Integrating with the existing LDAP database
seems like much more of a challenge.

I did find
http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html
and
http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/
very helpful.

For security reasons, we keep internal UIDs different from public email
IDs. Thus, we might use john.doe internally and j...@example.com for
email. Since it is a multi-tenant environment, I'd imagine we will use
the Kamailio domain module, make the SIP domain match the email domain,
and use the email user portion of the email address as the SIP ID. I
think this is straightforward using LDAP and Kamailio as we would query
LDAP for the email address and have return the password.

Asterisk seems a little trickier. I've looked at the schema extensions
and it looks like we add an auxiliary objectclass of AstSIPUser. I
suppose we would add this objectclass to a structure inetOrgPerson
object. We could then use the email name for the AstAccountName (or
whatever the actual attribute is) but the password befuddles me.

I notice we add an AstAccountRealmedPassword attribute. I suppose this
is because of the need to furnish SIP a hash derived from
username:realm:password. We would prefer our users only need to change
their passwords in one place. Is there anyway beside deploying
something like IPA to have Asterisk use the regular posix password
stored in LDAP rather than a separate AstAccountRealmedPassword?

I'm looking forward to diving in; I just wish it was with a little less
time pressure! Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

It also depends where you are registering your users. If merely using
Asterisk for a media server, do the auth via LDAP in Kamailio, which
will just use the userPassword attribute (or however the Kamailio LDAP
module binds to check auth or what you script it to do) then a normal
password change will do.

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

Sorry, lastly I defined it as auxilary to do exactly that; add it to
any existing entry.

Thanks.

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

One last thing ;-) use OpenLDAP!

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
 Most of the desktops are KDE and they use the KDE change password
 facility.  It works via pam I believe.  Is there an Asterisk interface
 with pam that would cause it to simultaneously change the Asterisk SIP
 realm password? If there is, I wonder how we pass it the requisite
 information? Thanks - John

No, but you could write one. You never mentioned how Asterisk is used
with Kamailio?

http://search.cpan.org/~nikip/Authen-PAM-0.16/d/PAM.pm



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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
 grin OpenLDAP isn't an option. And thanks very much for all the
 responses.  I've not had a chance to mock it up yet and see how it works
 hands on.  I am planning that the users ultimately interface SIP to
 Kamailio and use Asterisk for the call tree, voice mail, conference,
 etc.  I was assuming they would need to authenticate to Asterisk as well
 as Kamailio but I suppose it may be more a matter of Asterisk trusting
 Kamailio rather than the individual users.  I would also assume voice
 mail passwords will be very different from user passwords as they should
 be designed to be entered from a phone keypad rather than a keyboard (I
 told you I'm a real Asterisk newbie!).  I guess I'll find out as I start
 to set it up.

OK, depends how you set it up. You might not authenticate at all like
some ITSPs do (based on IP). Is this for your company?

 I committed a patch for voicemail passwords in the Asterisk LDAP
schema last week, so you'll need svn for that:

https://issues.asterisk.org/view.php?id=15155



 As I want to build it piecemeal and add complexity rather than diving
 into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with
 interaction as described above), any suggestions on whether I should
 build and test Kamailio or Asterisk first? Thanks - John

So, Asterisk and FreePBX? Why both?

This is a mighty big pie to take a bite out of, so it doesn't really
matter. Kamailio is harder is you don't know SIP. Depends, depends,
depends ;-)

What is the overall project goal here? We should have asked that first.

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Re: [asterisk-users] Realtime LDAP passwords

2009-06-02 Thread Gavin Henry

2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
 Thanks.  I do appreciate the input as I am jumping into the deep end as
 I said :)

 On Tue, 2009-06-02 at 21:43 +0100, Gavin Henry wrote:
 2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com:
  grin OpenLDAP isn't an option. And thanks very much for all the
  responses.  I've not had a chance to mock it up yet and see how it works
  hands on.  I am planning that the users ultimately interface SIP to
  Kamailio and use Asterisk for the call tree, voice mail, conference,
  etc.  I was assuming they would need to authenticate to Asterisk as well
  as Kamailio but I suppose it may be more a matter of Asterisk trusting
  Kamailio rather than the individual users.  I would also assume voice
  mail passwords will be very different from user passwords as they should
  be designed to be entered from a phone keypad rather than a keyboard (I
  told you I'm a real Asterisk newbie!).  I guess I'll find out as I start
  to set it up.

 OK, depends how you set it up. You might not authenticate at all like
 some ITSPs do (based on IP). Is this for your company?
 We are launching a new company whose primary product is a complete,
 hosted, virtualized environment including desktops for micro-businesses,
 charitable organizations, schools, and municipalities.  Unexpectedly,
 though not surprisingly, our initial customers are asking for a VoIP
 solution utilizing the same infrastructure. Hence the plunge into VoIP.
 We will be contracting with an ITSP for SIP trunking into our data
 center and need to set up the whole shooting match.

OK, to be honest then, since it's for a commercial solution and you're
so new, I'd buy something.

I've seen:

http://www.sipwise.com/index.php/products?start=2
http://www.asipto.com/
http://www.voice-system.ro/

I prefer the last one, but all vary on price and the money spent will
be saved on your dev time and learning curve. Then send yourself to
the training course. That way you know all the loop holes are closed
to allowing fraudulent calls etc.


  I committed a patch for voicemail passwords in the Asterisk LDAP
 schema last week, so you'll need svn for that:

 https://issues.asterisk.org/view.php?id=15155



  As I want to build it piecemeal and add complexity rather than diving
  into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with
  interaction as described above), any suggestions on whether I should
  build and test Kamailio or Asterisk first? Thanks - John

 So, Asterisk and FreePBX? Why both?
 From looking at the press release for AsteriskNOW (which I don't plan to
 use as I'd like a little tighter control over the system), it appears
 FreePBX and Asterisk 1.6 are a nice pairing and might ease some of our
 administration.  Just going on what I'm reading and not experience.


Sorry, I thought I read FreeSWITCH!

 This is a mighty big pie to take a bite out of, so it doesn't really
 matter. Kamailio is harder is you don't know SIP. Depends, depends,
 depends ;-)
 I'm reasonably comfortable with protocols and how they work (my
 background is as a network engineer although the skills are a bit
 rusty).  SIP seems quite comprehensible and all the docs I read through
 the night on the innards of Kamailio and SER made perfect sense.

 What is the overall project goal here? We should have asked that first.

 In effect, we will become a voice aggregator for micro-businesses and a
 shared PBX services provider to complement our data offerings. I was
 going to build Asterisk first to have complete standalone functionality
 but, if the user authentication will be primarily to Kamailio, it may
 make sense to start there.  I'll probably circle the pool a few times
 and then jump in wherever I stop unless someone with more experiences
 advises specifically! Thanks again - John
 --
 John A. Sullivan III
 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com

 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society


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Re: [asterisk-users] From 1.4 to 1.6.0

2009-05-19 Thread Gavin Henry

Is there any document on the reasons for the 1.6.0 and 1.6.1 branches?
I remember reading something but can't find it again.

Was it stability versus new features?

I'm currently playing with 1.6.1

Gavin.

On 19/05/2009, Benny Amorsen benny+use...@amorsen.dk wrote:
 Miguel Molina mmol...@millenium.com.co writes:

 Hi everyone,

 I was just wondering, does anyone managing production asterisk servers
 migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if
 there are some successful cases. Is your 1.6.0.X behaving well, with
 acceptable stability? Please share your experiences.

 Asterisk 1.6.0.5 and 1.6.1.0 are performing acceptably here. The new
 T.38 features are great, and the BLF/hint changes are nice too.

 This bug isn't so nice: https://issues.asterisk.org/view.php?id=13623
 We are currently trying the session-timers=refuse workaround.


 /Benny


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Re: [asterisk-users] Proxying from one server to another

2009-05-13 Thread Gavin Henry

Why not use OpenSIPS or Kamailio in stateful mode?

You will need to look at how media is handled though, but a SIP proxy
will work easily.

On 13/05/2009, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
 Hi David,



 Thanks for the reply. That's pretty much what I've already tried, but
 with no luck on the production machines.  In testing it worked, but the
 public IPs and single NICs were causing issues (we believe)

 So I was looking for a proxy-type solution.



 Adrian



 

 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
 Gibbons
 Sent: 13 May 2009 15:37
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Proxying from one server to another



 Redirect traffic with iptables like this:



 Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
 NEW_PUBLIC_IP



 I'm not sure if this will work for SIP. You may need the proxy to change
 info in the sip messages between server and client.



 --Dave





 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
 Marsh
 Sent: Wednesday, May 13, 2009 8:55 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Proxying from one server to another



 Hi All,



 I'm trying to find a software package to do the following sip proxy
 work:



 I've an A*k server A that needs to be decommissioned, from the USA, and
 replaced by server B, in the UK. Both servers are on public internet
 IPs.

 Whilst the client migration happens, I want to divert all the Register
 traffic from Server A to Server B to catch any clients still left out
 there.



 Unfortunately, the original Clients were configured with static IPs
 instead of DNS names for the SIP Registrar, so I have to proxy Server A
 until all the clients have been updated (which might be a long time).



 Obviously A*k itself wont do this (as far as I know).  I've looked at
 siproxyd and party-sip, but with no success so far.

 I've also tried using IPtables to redirect at the IP level, but the
 public IP ranges seem to stop me from achieving this. It works in my
 local-lan testing, but not on the public servers.



 Any ideas?



 Thanks,



 Adrian



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Re: [asterisk-users] Can someone help me with my IAX-registration

2009-05-04 Thread Gavin Henry

Is your box on a public ip or via nat? If eth0 isn't the ip you set it
to bind on it will ignore it.

I mean, is your * box on an internal address?

On 02/05/2009, jonas kellens jonas.kell...@telenet.be wrote:
 I have connected my Asterisk-box directly to my internetconnection. I
 have disabled my firewall.

 Still I am unable to register with my IAX-provider. Can someone please
 point me out why I am unable to register my Asterisk to another
 Asterisk-box ?
 A RegReq is send to the other Asterisk-box but no reply is received...
 No confirm, no reject...

 I have tried yet several configuration in my iax.conf file.
 There is also always the ignoring bindaddr and ignoring bindport...
 which doesn't sound right.

 What I know from books is that when you want to connect your Asterisk to
 another Asterisk-box you need to describe this other Asterisk-box as a
 peer.
 So my iax.conf :

 [general]
 autokill=yes
 bindport=4569
 bindaddr=78.22.166.226 ; the IP-address I get from my ISP

 register = cstore:my-passw...@ip-of-other-iax-box

 [cstore]
 type=user
 trunk=yes
 context=from-other-iax-box
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm

 [attractel] ; name I use in extensions.conf to contact other iax-box
 type=peer
 host=ip-address
 username=cstore ; username @ remote asterisk
 secret=my-password ; pass @ remote Asterisk
 auth=plaintext,md5
 trunk=yes
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm

 IAX reload :

 asterisk*CLI iax2 reload
   == Parsing '/etc/asterisk/iax.conf': Found
 [May  2 10:34:23] NOTICE[4626]: chan_iax2.c:10124 set_config: Ignoring
 bindport on reload
 [May  2 10:34:23] NOTICE[4626]: chan_iax2.c:10183 set_config: Ignoring
 bindaddr on reload
 doing dnsmgr_lookup for '62.213.196.38'
   == Parsing '/etc/asterisk/users.conf': Found
 doing dnsmgr_lookup for '62.213.196.38'
   == Loaded firmware 'iaxy.bin'
   == Parsing '/etc/asterisk/iaxprov.conf': Found
 -- Loaded provisioning template 'default'

 Is ignoring bindaddr and bindport normal ?

 IAX debug :

 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
Timestamp: 00019ms  SCall: 13117  DCall: 0 [62.213.196.38:4569]
USERNAME: cstore
REFRESH : 60

 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
Timestamp: 00019ms  SCall: 13117  DCall: 0 [62.213.196.38:4569]
USERNAME: cstore
REFRESH : 60

 Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
Timestamp: 00019ms  SCall: 13117  DCall: 0 [62.213.196.38:4569]
USERNAME: cstore
REFRESH : 60

 IAX status :

 asterisk*CLI iax2 show registry
 Host   dnsmgr  UsernamePerceived
 Refresh   State
 62.213.196.38:4569N   cstore  Unregistered   60
 Request Sent

 Host   dnsmgr  UsernamePerceived
 Refresh  State
 62.213.196.38:4569N   cstore  Unregistered   60
 Timeout


 Thanks for the help,
 Jonas.


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Re: [asterisk-users] Jabber and Presence

2009-04-24 Thread Gavin Henry

2009/4/23 Matt Riddell li...@venturevoip.com:
 On 18/04/2009 2:28 a.m., Gavin Henry wrote:
 Hi all,

 What other open source tools are people using for this? I was looking
 at Openfire and their asterisk plugin.

 Is it easy to roll your own with res_jabber.so ??

 I used openfire in the past, but have now changed over to using ejabberd.

 We use PHP classes to send jabber messages from the support system,
 JabberSend to send messages from the dialplan, and a bot to send
 messages for live support.

Thanks for that Matt

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Re: [asterisk-users] Zaptel to Dahdi

2009-04-20 Thread Gavin Henry

2009/4/20 jonas kellens jonas.kell...@telenet.be:
 Please, is there anyone who can help me with this zaptel -- Dahdi -problem
 ??

 Will chan_dahdi.conf work with zaptel.conf ? Will Asterisk be able to
 communicate with the Digium TDM pci-card ?

 Or do I need to compile dahdi and recompile Asterisk ???

 Thank you for your reply.

Why not just try it.

 Jonas.


  Forwarded Message 

 From: jonas kellens jonas.kell...@telenet.be
 To: asterisk-users@lists.digium.com
 Subject: Zaptel to Dahdi
 Date: Sun, 19 Apr 2009 17:17:39 +0200

 VoIP-wiki.org states :

 /etc/zaptel.conf Becomes /etc/dahdi/system.conf
 /etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.conf

 Now, what do I have installed on my system :

 /etc/zaptel.conf and /etc/asterisk/chan_dahdi.conf

 Will these two config-files work together ???

 I have no /etc/asterisk/zapata.conf and no /etc/dahdi/system.conf

 Do I create an empty zapata.conf ??

 I also do not have /usr/lib/asterisk/modules/chan_zap.so !!

 My Asterisk-version : 1.4.24

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[asterisk-users] Jabber and Presence

2009-04-17 Thread Gavin Henry

Hi all,

What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.

Is it easy to roll your own with res_jabber.so ??

Thanks.

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[asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

Hi all,

Has anyone put * in between an Avaya and Cisco system to connect two
offices together?

I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them. There is a leased line for
bandwidth.

Any tips/ideas on whether this is possible or dumb?

Thanks.

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Re: [asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

BTW, what's the recommended production version of Asterisk source
you'd recommend, the latest 1.4 or 1.6?

In fact, nevermind. This is asked so many times I'll hit the archives.

Cheers.

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Re: [asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

2009/4/3 John Todd jt...@digium.com:

 On Apr 3, 2009, at 7:40 AM, Gavin Henry wrote:

 Hi all,

 Has anyone put * in between an Avaya and Cisco system to connect two
 offices together?

 I was thinking about adding a SIP trunk on each side and getting
 Asterisk to pass calls between them. There is a leased line for
 bandwidth.

 Any tips/ideas on whether this is possible or dumb?

 Thanks.


 Gavin -
   The short answer is yes, this is possible, and is done quite
 often.  How exactly you configure it is of course the trick - there
 are many possible different methods by which you might accomplish this
 feat, depending on what your existing resources are and what your end
 goal is.  T1? PRI? H.323?  You may consider IAX2 for trunking and save
 a lot of bandwidth as compared to SIP, if bandwidth is a concern.  If
 you're using T1 or PRI, you'll need a hardware card to do this.

   I'd start with setting up a basic Asterisk server from source and
 getting two SIP phones working on it.  I'd not suggest using one of
 the GUI-enabled versions - that may be more layers of stuff than
 you're looking for given your stated goal.  Figure it out, read the
 O'Reilly Book (Asterisk: The Future of Telephony) and you'll probably
 figure out fairly quickly how to use Asterisk as a black-box trunking
 interface for your systems.

Thanks John. Yeah, we've done this for an Avaya system already using
H.323 and we can
just add a sip trunk to the CCM and do dialplans accordingly. Just
need to get some specs on
what each side is from the client.

We could put a simple box on each side and use IAX2 trunking, sure.

It's simple and I should have thought it through before posting ;-)

Cheers John.

Gavin.

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Re: [asterisk-users] Multi-tenant with receptionist features for managed service

2009-03-17 Thread Gavin Henry

2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
 On Mon, 16 Mar 2009, Gavin Henry wrote:

 Dear all,

 I'm currently researching options for a MT asterisk gui/system for a
 small business centre that will have 12 units in it. Each unit will be
 configured for one extension.

 The system there will have a max of 12 concurrent calls to PSTN
 provided via an ADSL/SDSL link to our VoIP provider in the UK, using
 g.711, maybe g.729 dependant on networking costs. Fallback will
 be to 4 analogue lines should this go down.

 Gavin,

 You won't get 12 concurent G711 calls over a standard ADSL line in the UK.
 If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but
 even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or
 get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will
 give you a few extra channels though as the IP overhead is less.

Thanks. We're waiting to hear abou twhat we can provide. We use Gradwell for
termination and their ADSL. DSL Premium M does 2.5 up, but I'll limit
this to 10 calls
to be safe.

 What is key is billing information and the ability for a receptionist
 to see all active calls and do transfers etc. Much like the Flash
 Operator Panel. Desktop Software may also be needed for this purpose
 or can be done via a traditional bank of lines on an IP phone
 accessory module.

 Have a look at: http://www.astassistant.com/ rather than FOP. Even has a
 Linux client which is nice...

Looks good. Just tested it on VirtualBox for box.

 If anyone has any ideas on the best way to put this together, I'm all ears
 ;-)

 The consultant in me says Pay someone to do it for you :) However it's not
 that hard to do and setup if youve done something similar in the past - and
 your budget is tight. If you know you're going to get more of these, then go
 for it - spend your time on the software and front-end for the the first
 one, then the rest are clones...

Yeah. I normal use PBXinAFlash for this. Just the receptionist part
that was missing
and maybe add on a2billing.

 I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra
 53i phones. There's a £4k budget for this (still waiting for more
 into)which
 will include the networking connection and equipment. If I can afford it I
 normally go Sangoma with Echo cancellation, but as it's a fallback
 service,
 so I'm not bothered.

 When budgets tight - I've deployed a lot of Grandstream phones - might give
 you a bit more breathing space if you use (eg) GXP280's for the client
 phones and a GXP2000 + button box for the receptionist.

Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT.

 You can save money by building your own hardware too. Atom mobo, 1GB of RAM
 and an OpenVox card running oslec is still overkill for this. I mostly use
 1GHz VIA boards for these sort of projects with up to 60 extensions.

What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and
a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday.

 A 4 port FXO card is £126.95 ex vat.

 Billings a PITA and other than what I've written myself, have never found
 anything that works the way I'm happy with... Good luck!

Thanks.

 I think I've covered everything. There will be many more business
 centres to come as this first project will be the blueprint one. The
 end goal is to also move this to a data centre and not have it on site
 with the pstn fallback options, but use redundant links to our DC.
 Like a mini-ITSP for our area. I haven't figured the receptionist part
 for that bit yet though ;-)

 Personally I'd stick the box on-site and have a central peering server or 2
 in the DC - well that's how I do it ;-) You'll struggle to get properly
 redundant links in that budget range too - one JCB can ruin everyones day!

Yeah, as I planned, but not for this project.

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Re: [asterisk-users] Multi-tenant with receptionist features for managed service

2009-03-17 Thread Gavin Henry

2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
 On Tue, 17 Mar 2009, Geraint Lee wrote:

 We can put about 9/10 calls using SIP/gsm through our BT Business Network
 ADSL package connection (832kbit upstream, £65/month) before you notice
 the
 quality starting to drop, but you could always get two connections and
 bond them together into one using openvpn or some other method if you
 wanted to.

 Ugh. GSM )-:

 I've never really had much luck with BT as an Internet provider either -
 their wholesale network - good, retail broadband, bad...

 In theory, you should be able to get 10 G711 SIP calls over a business
 quality 830Kb/sec upload ADSL line. I get 9 on my test setup before any
 packet loss. I managed 11 calls using IAX over the same line before loss.
 (Entanet ADSL and a Draytek router - £25 a month)

 Intersting idea re. using openvpn or similar.. I have sites with 3 ADSL
 connections - one for incoming calls, one for outgoing and one for general
 office use.. That works when the call numbers in/out is relatively balanced
 though.

 I know of a local company who're regularly putting 20 concurrent calls over
 the same broadband setup using G729...

Yeah, we use g.729 ourselves too.

Gavin.

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Re: [asterisk-users] Multi-tenant with receptionist features for managed service

2009-03-17 Thread Gavin Henry

A2billing is a good fit for that then. Yeah, voipon. Thanks for the
input Gordon. Maybe worth hooking up offline if we're doing similar
stuff.

Gavin.

On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
 On Tue, 17 Mar 2009, Gavin Henry wrote:

 2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
 On Mon, 16 Mar 2009, Gavin Henry wrote:

 When budgets tight - I've deployed a lot of Grandstream phones - might
 give
 you a bit more breathing space if you use (eg) GXP280's for the client
 phones and a GXP2000 + button box for the receptionist.

 Yeah, don't really like them though. I could go down to a 51i for £67 ex
 VAT.

 Grandstreams aren't to everyones liking, this is true...

 You can save money by building your own hardware too. Atom mobo, 1GB of
 RAM
 and an OpenVox card running oslec is still overkill for this. I mostly
 use
 1GHz VIA boards for these sort of projects with up to 60 extensions.

 What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM
 and
 a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday.

 Under £200 from someone like http://linitx.com/ I don't put disk drives in
 my boxes though - they boot out of flash. I guess with the Dell, you have
 on-site or next day replacement if you take that deal though.

 A 4 port FXO card is £126.95 ex vat.

 (From voipon by the looks of that price ;-)

 Billings a PITA and other than what I've written myself, have never found
 anything that works the way I'm happy with... Good luck!

 Thanks.

 I've been approcached by a client who wants a sort of hotel billing system
 though - tailored to their needs - it's for a retirement home sort of
 thing. I suggested they just did a fixed-price deal with the inmates, but
 that didn't go down well. They want to account for everything to the
 last penny )-:

 I think I've covered everything. There will be many more business
 centres to come as this first project will be the blueprint one. The
 end goal is to also move this to a data centre and not have it on site
 with the pstn fallback options, but use redundant links to our DC.
 Like a mini-ITSP for our area. I haven't figured the receptionist part
 for that bit yet though ;-)

 Personally I'd stick the box on-site and have a central peering server or
 2
 in the DC - well that's how I do it ;-) You'll struggle to get properly
 redundant links in that budget range too - one JCB can ruin everyones
 day!

 Yeah, as I planned, but not for this project.

 Good luck!

 Gordon


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Re: [asterisk-users] Multi-tenant with receptionist features for managed service

2009-03-17 Thread Gavin Henry

Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.

On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
 On Tue, 17 Mar 2009, Gavin Henry wrote:

 2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
 On Tue, 17 Mar 2009, Geraint Lee wrote:

 I know of a local company who're regularly putting 20 concurrent calls
 over
 the same broadband setup using G729...

 Yeah, we use g.729 ourselves too.

 The issues I've had have been when theres transcoding going on that you
 can't control - ie. outside your network, so I can go point to point from
 end-user phone to the people I peer with, but if they then transcode to
 G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for
 a mobile, or back to G729 to go to an expensive overseas location, then
 quality does suffer )-:

 Gordon

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[asterisk-users] Multi-tenant with receptionist features for managed service

2009-03-16 Thread Gavin Henry

Dear all,

I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.

The system there will have a max of 12 concurrent calls to PSTN
provided via an ADSL/SDSL link to our VoIP provider in the UK, using
g.711, maybe g.729 dependant on networking costs. Fallback will
be to 4 analogue lines should this go down.

What is key is billing information and the ability for a receptionist
to see all active calls and do transfers etc. Much like the Flash
Operator Panel. Desktop Software may also be needed for this purpose
or can be done via a traditional bank of lines on an IP phone
accessory module.

If anyone has any ideas on the best way to put this together, I'm all ears ;-)

I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra
53i phones. There's a £4k budget for this (still waiting for more into)which
will include the networking connection and equipment. If I can afford it I
normally go Sangoma with Echo cancellation, but as it's a fallback service,
so I'm not bothered.

I think I've covered everything. There will be many more business
centres to come as this first project will be the blueprint one. The
end goal is to also move this to a data centre and not have it on site
with the pstn fallback options, but use redundant links to our DC.
Like a mini-ITSP for our area. I haven't figured the receptionist part
for that bit yet though ;-)

Thanks,

Gavin.

P.S. I have thought about pbxinaflash and a2billing, but I'm not sure
if it would not be clunky for a novice to handle (receptionist). I may
go down that route and hire the FreePBX team to fill in the mixing pieces
of Multi-tenant if they are interested.

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Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

2009-03-13 Thread Gavin Henry

2009/3/12 Paulo Santos paulo.r.san...@sapo.pt:
 Gavin Henry wrote:
 Hi All,

 We've got msidn configured:

 Port  1: TE-mode BRI S/T interface line (for phone lines)
  - Protocol: DSS1 (Euro ISDN)
  - childcnt: 2
 

 I don't know if it depends on the card, but in my case I need to set the
 termination jumper on TE mode for lines from PSTN. Mind to check the
 TE/NT jumper as well.


 te_ptmp=1

 (...)

 [isdn]
 ports=1
 context=from-pstn
 msns=*

Everything worked first time, so thanks!

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Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

2009-03-13 Thread Gavin Henry

2009/3/12 Giorgio Incantalupo gincantal...@fgasoftware.com:
 Hi Gavin,

 if you can make and receive calls it works...do not worry if your line
 is shown as DOWN, some telco turns it off but it works without problem.
 Remember to ask your telco for the right signalling and set it the right
 way (PTP or PMP).

Thanks. It's all working with above, I just hadn't tested an inbound
call. Pretty lucky really ;-)

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[asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

2009-03-12 Thread Gavin Henry

Hi All,

We've got msidn configured:

Port  1: TE-mode BRI S/T interface line (for phone lines)
 - Protocol: DSS1 (Euro ISDN)
 - childcnt: 2


mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
iend(0x8fd5060)


 and running on Asterisk 1.4.21.2:

pbx*CLI misdn show stacks
BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0  Debug:0

but I'm not sure how to check our settings are right with Etisalat, as
above then goes to:

pbx*CLI misdn show stacks
BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0

/etc/misdn-init.conf:

card=1,hfcpci
te_ptmp=1
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0

/etc/asterisk/misdn.conf:

[general]
debug = 0
method=standard
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=misdn
language=en
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
callgroup=1
pickupgroup=1
presentation=-1
screen=-1
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no

[isdn]
ports=1
context=from-pstn
msns=*


Now we've setup PRi before, but not BRI and not in Dubai.

If anyone has an idea or config that would be great and I'd make sure
it goes on voip-info.org for others too.

Thanks,
Gavin.

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Re: [asterisk-users] Simple Meetme Question

2009-03-08 Thread Gavin Henry

Just transfer them to your meetme extension after you've called them.
Just like you would transfer someone who has called you.

* will then put them into that conference.

Thanks.

On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote:
 Hello,

 setting up Meetme was very easy. I jut added the MeetMe Application to
 an internal extension to be reachable by SIP and to an external
 extension to be reachable by ISDN.

 What I don't understand however is how to call somebody and drop him
 to the conference?

 I'm using Asterisk 1.4 from Debian lenny

 Sven

 --
 In the land of the brave and the free, we defend our freedom
 with the GNU GPL (Richard M. Stallman on www.gnu.org)

 /me is gig...@ircnet, http://sven.gegg.us/ on the Web


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Re: [asterisk-users] Current state of Asterisk and Virtualization?

2009-02-27 Thread Gavin Henry

2009/2/27 John Todd jt...@digium.com:

 On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote:

 Gavin Henry wrote:
 Hi all,

 In a pure VoIP env, what is the current state of do's and don't s of
 virtualizing * in order to provide multiple separate instances, say
 for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?

 I've read lots of threads going back to 2007 and I'm in the general
 option that kvm is the way to go now, if at all.

 If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a
 conference box could be put along side the vm hardware and have a
 card
 in it.

 Thoughts, experiences and being told to shut up are all very much
 appreciated.

 Thanks.


 http://www.bicomsystems.com/products/C/P/797/411/



 ...and also:

 http://voxilla.com/2009/02/12/amazon-ec2-voip-1096
 http://voxilla.com/2009/02/13/asterisk-amazon-ec2-1178
 http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405

 and:

 http://www.simionovich.com/?p=180
 http://www.simionovich.com/?p=243
 ...and lots of others on Nir's blog.


 Even MORE resources/questions/answers:

 http://www.google.com/search?hl=enq=ztxenbtnG=Google+Searchaq=foq=
 http://www.google.com/search?num=30hl=ensafe=offq=xen+and+ztdummybtnG=Search

 JT


 ---
 John Todd                       email:jt...@digium.com
 Digium, Inc. | Asterisk Open Source Community Director
 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
 direct: +1-256-428-6083         http://www.digium.com/




Great, links. Will be back with comments/questions later.

Thanks.

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[asterisk-users] Current state of Asterisk and Virtualization?

2009-02-26 Thread Gavin Henry

Hi all,

In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?

I've read lots of threads going back to 2007 and I'm in the general
option that kvm is the way to go now, if at all.

If dadhi_dummy/zt_dummy is still an issue for conferencing etc. a
conference box could be put along side the vm hardware and have a card
in it.

Thoughts, experiences and being told to shut up are all very much appreciated.

Thanks.

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Re: [asterisk-users] Asterisk 1.4 and openLDAP

2008-10-19 Thread Gavin Henry

That looks cool. Will have a play.

On 10/18/08, Ming Yong [EMAIL PROTECTED] wrote:
 Anael,
 You should take a look at Druid (Open Source Unified Communications)
 Project based on Asterisk that has complete LDAP backend and Zimbra
 connector.
 It's an open source project  we are looking for collaborators  users.

 Druid UCS 5.0 with LDAP backend
 http://www.youtube.com/watch?v=Xl78orka938

 Druid Zimlet for click to call and drag  drop faxing
 http://www.youtube.com/watch?v=WdEVSJuh1ow

 Ming

 On Sat, Oct 18, 2008 at 3:30 PM, Anael DIAZ [EMAIL PROTECTED] wrote:
 Hi there,

 I need help in implementing Asterisk with LDAP. I' ve installed Asterik
 1.4
 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD.

 thanks,




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 --
 Ming Yong
 CEO, www.voiceroute.org
 Druid - Open Source Unified Communications
 DID: +1-877-242-3704
 Office: +1-866-915-2407 ext 301
 SIP/email: [EMAIL PROTECTED]
 --
 VoiceCON 08 San Francisco
 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA
 http://druidvoicecon.eventbrite.com

 Voiceroute videos on Druid, Open Source Unified Communications  Asterisk
 http://youtube.com/voiceroute

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Re: [asterisk-users] Asterisk 1.4 and openLDAP

2008-10-19 Thread Gavin Henry

The LDIF needs updating as it's not a working example. I'll have one
next week. I'll release an updated schema too.

Gavin.

On 10/18/08, Tilghman Lesher [EMAIL PROTECTED] wrote:
 On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote:
 I need help in implementing Asterisk with LDAP. I' ve installed Asterik
 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra
 LPAD.

 You might want to take a look at Asterisk 1.6, which has LDAP realtime
 support.  Look within contrib/scripts to find a working example of an LDIF
 and schema file for use with Asterisk.

 --
 Tilghman

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Re: [asterisk-users] Global VoIP Calls?

2008-08-25 Thread Gavin Henry

 Or provide both solutions - let the offices call each other via VoIP, but
 if too laggy, fall-back to VoIP - PSTN... (- VoIP)

How can you test for this precall?

Cheers.

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Re: [asterisk-users] Global VoIP Calls?

2008-08-24 Thread Gavin Henry

Thanks all for your suggestions.

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[asterisk-users] Global VoIP Calls?

2008-08-23 Thread Gavin Henry

Dear All,

What setup would you recommend for making VoIP calls whilst bringing
latency down between offices at:

* Edinburgh
* Kuala Lumpur
* Singapore
* Tokyo
* Seoul
* Beijing
* San Francisco

Some of the Asia offices are  300ms some  200ms.

Any advice greatly apreciated.

Thanks.

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Re: [asterisk-users] Call quality

2008-07-02 Thread Gavin Henry

2008/7/2 Loic Didelot [EMAIL PROTECTED]:
 Depends on the phone.

 On many devices you can setup buttons to call a url. Thats what I did.

Ah, yes. Would be a good thing to implement here. Then you can do
anything, like a support ticket etc.

Cheers.

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Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?

2008-07-02 Thread Gavin Henry

We do as do Gradwell.com

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Re: [asterisk-users] Call quality

2008-07-01 Thread Gavin Henry

What did you do to setup a button for alerts?

Thanks.

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Re: [asterisk-users] Google Apps IMAP

2008-06-25 Thread Gavin Henry

Google Apps version might.

2008/6/25 Marc Smith [EMAIL PROTECTED]:
 Hi,

 Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
 IMAP? If so, does their IMAP implementation support any kind of
 master user (Dovecot) abililty? Good? Bad?

 --Marc

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Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)

2008-06-24 Thread Gavin Henry

Did I add this yet?

2008/3/22 Faraz Khan [EMAIL PROTECTED]:
 Just checked 1.6.0beta4 - the res_ldap.conf file still has PBX*
 attributes - which I'm guessing would be confusing to any new user.

 the schema file looks file though, the missing voicemail/queue part is
 what we have added.


 Quoting Faraz Khan [EMAIL PROTECTED]:

 Did you manage to upload those changes? Some of your schema/ldif files
 were deleted by the bug admin. You might want to upload them at
 voip-info

 Furthermore, the multi_ldap call is broken in res_config_ldap.c - I
 even started a bounty on it but looks like few people are interested
 and/or bounty amount is too low :)

 without the multi_ldap fix, all we can realistically do is put
 sip.conf in ldap- which is a decent improvement however it would be
 amazing if the entire dialplan/queues/etc could be put into voicemail
 as well. Right now one has to use LDAP for account and Mysql for
 extensions/queues.

 Quoting Gavin Henry [EMAIL PROTECTED]:

 On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
 Good Idea and done. It is now available here:

  http://www.voip-info.org/wiki/view/LDAP

 The correct LDAP Schema is included:

 /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap-schema

 and

 /asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldif

 Good work though. I'm just uploading some fixes to it at:

 http://bugs.digium.com/view.php?id=12177

 Gavin.

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 --
 Faraz R Khan
 Chief Architect
 Emergen Consulting Pvt Ltd
 www.emergen.biz

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 --
 Faraz R Khan
 Chief Architect
 Emergen Consulting Pvt Ltd
 www.emergen.biz

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Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.

2008-06-17 Thread Gavin Henry

2008/6/16 Syed Nasruddin [EMAIL PROTECTED]:


 Thanks for the link. I think I will be using this product.

It's very, very good. You can hook it up to MySQL instead of sqlite if
needed, just e-mail support.

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Re: [asterisk-users] cdr-custom/Master.csv rotation

2008-06-15 Thread Gavin Henry

2008/6/15 Mark Hamilton [EMAIL PROTECTED]:
 Yup, drive. Or in Gavin's case Fly.
 Really appreciate your help, Darryl. Thanks a lot.

Sorry, sometime I presume people just need a pointer in the right direction.

I should have said have a look at /etc/logrotate.* files on a
GNU/Linux box and copy one.

Then if you got stuck I coudl have done a copy and paste for you.

 I'm attempting to use this now as is, but Tzafrir points out that this might
 not rotate Master.csv in /cdr-custom. In such a case, what would I need to
 do?

 Thanks again,
 Mark.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin
 Sent: June 14, 2008 11:49 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation

 It's like asking for directions, and someone tells you to drive,
 useless.

 Here is what we do here:
 Create /etc/logrotate.d/asterisk:
 /var/log/asterisk/asterisk-verbose /var/log/asterisk/messages
 /var/log/asterisk/debug /var/log/asterisk/queue_log {
daily
rotate 7
compress
missingok
notifempty
sharedscripts
postrotate
/usr/local/bin/log_rot_ast
endscript
 }

 /usr/local/bin/log_rot_ast contains:
 #!/bin/sh
 /usr/sbin/asterisk -rx 'logger reload' /dev/null 21

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mark
 Hamilton
 Sent: Saturday, June 14, 2008 19:05
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation

 Gavin,


 I really do appreciate your one-liner. But is there any more insight
 into
 this? I know I have to use Logrotate, but I have no idea how I can
 actually
 get it done.

 I'm going to try and figure it out right now, but for the benefit of the
 list and archives, it just might be good if solutions could be posted
 here
 too.

 Thanks,
 Mark.

 PS: Remember, many people get their answers from mailing list archives.
 So
 we'd rather get them solved than getting the same question on the list 3
 months later. :)


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Gavin
 Henry
 Sent: June 13, 2008 4:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation

 2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
 Hi,



 How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by
 date?


 Logrotate on a *nix box.

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Re: [asterisk-users] cdr-custom/Master.csv rotation

2008-06-15 Thread Gavin Henry

2008/6/15 Darryl Dunkin [EMAIL PROTECTED]:
 It's like asking for directions, and someone tells you to drive,
 useless.

It's not useles. What will he learn by just copying and pasting below?

The first thing I would have done if I got a reply that said just
logrotate is Google for it and then read it's man page.

 Here is what we do here:
 Create /etc/logrotate.d/asterisk:
 /var/log/asterisk/asterisk-verbose /var/log/asterisk/messages
 /var/log/asterisk/debug /var/log/asterisk/queue_log {
daily
rotate 7
compress
missingok
notifempty
sharedscripts
postrotate
/usr/local/bin/log_rot_ast
endscript
 }

 /usr/local/bin/log_rot_ast contains:
 #!/bin/sh
 /usr/sbin/asterisk -rx 'logger reload' /dev/null 21

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mark
 Hamilton
 Sent: Saturday, June 14, 2008 19:05
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation

 Gavin,


 I really do appreciate your one-liner. But is there any more insight
 into
 this? I know I have to use Logrotate, but I have no idea how I can
 actually
 get it done.

 I'm going to try and figure it out right now, but for the benefit of the
 list and archives, it just might be good if solutions could be posted
 here
 too.

 Thanks,
 Mark.

 PS: Remember, many people get their answers from mailing list archives.
 So
 we'd rather get them solved than getting the same question on the list 3
 months later. :)


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Gavin
 Henry
 Sent: June 13, 2008 4:51 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation

 2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
 Hi,



 How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by
 date?


 Logrotate on a *nix box.

 --
 http://www.suretecsystems.com/services/openldap/

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Re: [asterisk-users] Using Asterisk Only as Voice Recording Solution.

2008-06-13 Thread Gavin Henry

2008/6/12 Syed Nasruddin [EMAIL PROTECTED]:


 HI,



 I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair
 command over Asterisk up till now and have run it in different scenarios
 such as Call Center Solution, PBX solution.



 There is a requirement to use Asterisk only as Voice Recording solution in
 following manner:



 Physical POT lines before entering into our native PBX will be splitted and
 one of each of those lines will also enter into our Asterisk System.
 Once the call is routed by our native PBX and recipient picks up the phone
 (either SIP phone or Analog Phone) I should be able to start recording the
 call.
 When the call ends, the recording should stop.

Our clients use this for E1 Pri: http://www.voicetronix.com/logger.htm

Not sure if there is a analogue solution.

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Re: [asterisk-users] cdr-custom/Master.csv rotation

2008-06-13 Thread Gavin Henry

2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
 Hi,



 How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date?


Logrotate on a *nix box.

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[asterisk-users] Asterisk can handle only 200 to 300 SIP device registrations

2008-06-08 Thread Gavin Henry

Hi All,

Is this still the cause in 1.4 and 1.6 as per:

http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.

Do people recommend OpenSER in front for deployments bigger than 300 end points?

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

2008-06-07 Thread Gavin Henry

What model in the Polycom or Aastra range is the 360 level with?

2008/6/6 Chris Bagnall [EMAIL PROTECTED]:
 When I pushed some vendors for prices there was only a tiny gap between
 the 300 and 360.  Would suggest looking hard at the 360 always...

 Interesting... here in the UK the price difference between the 300 and 360 is 
 pretty huge. Either you're getting some stunningly good pricing on 360s or 
 some abysmal pricing on the 300s :-)

 Regards,

 Chris



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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

2008-06-07 Thread Gavin Henry

2008/6/7 Gavin Henry [EMAIL PROTECTED]:
 What model in the Polycom or Aastra range is the 360 level with?

Probably the IP601:

http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html

and 57i:

http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html

Snom 360:

http://www.voipon.co.uk/snom-360-ip-telephone-p-31.html

They all have the 12 keys.

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Re: [asterisk-users] Any reason to not use AEL? (Also, MixMonitor q)

2008-06-04 Thread Gavin Henry

What about using RealTime LDAP in 1.6? That woudl be much faster than a RDBMS.


2008/6/3 Sherwood McGowan [EMAIL PROTECTED]:
 Mindaugas Kezys wrote:
 Thank you for your opinion.

 Then my question would follow: how to build human-friendly system which will
 use GUI and lets user use that system without messing with .conf files?

 From my experience large and complicate systems can't be effectivelly
 managed without Realtime and I see no way how to put AEL into DB. Maybe it's
 possible?

 We are storing exact-match info into DB and all _X., etc stuff we have in
 extensions.conf. So no speed issues with large systems.

 Also: Any reason to not use extensions.conf?

 What AEL can do better then extensions.conf?

 Many people still use vi. Because it can do everything what they want. Same
 here with extensions.conf.

 Regards,
 Mindaugas Kezys
 http://www.kolmisoft.com



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Steve Murphy
 Sent: Tuesday, June 03, 2008 9:02 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Any reason to *not* use AEL? (Also,
 MixMonitor q)

 On Tue, 2008-06-03 at 09:33 -0500, Sherwood McGowan wrote:

 Mindaugas Kezys wrote:

 Does Asterisk Realtime support AEL?



 Regards,

 Mindaugas Kezys

 http://www.kolmisoft.com



 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of
 *Gonzalo Servat
 *Sent:* Tuesday, June 03, 2008 5:07 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Any reason to *not* use AEL? (Also,
 MixMonitor q)



 On Tue, Jun 3, 2008 at 10:41 AM, Eric Wieling [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 AEL in 1.4 was the first version of AEL that most people

 consider

 stable.  Since not many people uses AEL, you won't get nearly

 as

 much
 (if any) community support compared to if you are using the
 non-AEL syntax.


 Really? Why would anyone want to write a dialplan using the old
 extensions.conf syntax? That sort of syntax personally drove me

 nuts

 (and real messy). I've got my entire dialplan on AEL (using

 Asterisk

 1.6.0).


 -

 Not sure what you mean, but if you mean realtime dialplan, then no,
 you can't use AEL for that. However, we might wish to see if Murf
 knows if this can be done.


 extensions.conf is like assembler; it's a very strict, line per
 instruction format, 4 fields per line, that is able to be read in by
 normal config file parsers. It is in turn compiled into the internal
 asterisk data structures.

 AEL is more free form. Storing the dial plan in AEL format in a db
 would be pretty useless. However, the extensions.conf isn't so bad in a
 db, as it still has the 4 columns, row per instruction sort of format.

 But in general, I have to ask, as a programmer, if it's really, really
 a good idea to store code in a db. The dialplan is a mixture of both
 dialplan code and data, in the form of extensions.

 But storing dialplan code, as in a sequence of application calls, is
 a slow way to execute your dialplan code.

 And storing patterned extensions (extensions starting with _, like
 _10X or whatever), is a really slow way to match pattern
 extensions. My advise to everyone is this: Realtime is great, but don't
 store extension patterns in there, and don't store your dialplan code
 in there, if you can help it. It'd be much better to use your db to
 store 'exact' extension data. Trying to find the best pattern match via
 realtime is excruciatingly slow, as it calls up every extension in the
 db for that context, and then decides on the best match.  DB's do a
 great job at storing large numbers of uniquely keyed data that you can
 find via exact matches. So, use a general exten patten in your
 dialplan, and then do a DB() lookup from there.

 If you find a bug in your dialplan code, you've got to change it in two
 places, in the realtime db, and you'd best have it in your original
 source as well, in case you need to reload/recover your db or whatever.
 A DB is a lousy source-code control system. Use cvs or subversion or
 git or something to store your dialplan code instead. That way, you can
 back out change sets, etc, and track your changes in a much more
 practical way.

 Just my two cents.

 murf

 --
 Steve Murphy
 Software Developer
 Digium



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 There's not really a reason to NOT use extensions.conf, other than it's
 spaghetti code, and not as readable. You CAN write a gui that alters the
 *.ael files, that's how a lot of the GUIs work for extensions.conf, they
 modify the file. Putting your dialplan into a database is needless in
 about 90% of cases I've run across.

 --
 Sherwood

Re: [asterisk-users] Any reason to not use AEL? (Also, MixMonitor q)

2008-06-04 Thread Gavin Henry

2008/6/4 Tzafrir Cohen [EMAIL PROTECTED]:
 On Wed, Jun 04, 2008 at 10:45:13AM +0100, Gavin Henry wrote:
 What about using RealTime LDAP in 1.6? That woudl be much faster than a 
 RDBMS.

 If performance is such a major issue, why not use explicit queries?

 realtime has overhead even in extensions/proiorities where it is not used.

Static will always be faster than any Realtime. But as I understand
it, some things should be kept out of it.

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Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)

2008-03-18 Thread Gavin Henry

On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
 Good Idea and done. It is now available here:

  http://www.voip-info.org/wiki/view/LDAP

The correct LDAP Schema is included:

/asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap-schema

and

/asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldif

Good work though. I'm just uploading some fixes to it at:

http://bugs.digium.com/view.php?id=12177

Gavin.

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Re: [asterisk-users] LDAP support

2008-02-01 Thread Gavin Henry

There a realtime LDAP driver now in 1.6beta2

On 23/01/2008, Cavalera Claudio Luigi [EMAIL PROTECTED] wrote:
 Hello,
 I've found this information about asterisk and LDAP:
 http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
 which can be out of date.

 I'm trying this http://www.mezzo.net/asterisk/app_ldap.html
 however I'm facing the same problems as this unanswered:
 http://forums.digium.com/viewtopic.php?p=42591sid=05e1d00ab6f9848f4e7b6
 39d66cc6d79
 Does anybody know how to solve this issue?

 Moreover I would like to understand if someone is using LDAP (for
 iax.conf) and with which asterisk plugin (e.g. app_ldap,
 Asterisk::LDAP Perl module, etc..).

 Best Regards,
 Claudio


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Re: [asterisk-users] How long to detect an h exten?

2007-08-31 Thread Gavin Henry

On 30/08/2007, C F [EMAIL PROTECTED] wrote:
 Can you explain this question?
 Just to clearify, exten = h will execute as soon as Asterisk is aware
 that the channel was hung up. While app_hangup will execute a hangup
 on an active channel.

I'm just trying to track down some delays in my dialplans and wondered
if this might the problem in hanging up a zap call.



 On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote:
  Dear All,
 
  How long should it take before a exten = h,1,Hangup() kicks in,
  versus a exten = s,n,Hangup()
 
  I'm just about to test, but thought I'd ask.
 
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[asterisk-users] How long to detect an h exten?

2007-08-30 Thread Gavin Henry

Dear All,

How long should it take before a exten = h,1,Hangup() kicks in,
versus a exten = s,n,Hangup()

I'm just about to test, but thought I'd ask.

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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-29 Thread Gavin Henry

No probs.

On 29/08/2007, Abhishek M S [EMAIL PROTECTED] wrote:
 Dear Mr Gavin,
 Thank you once again. Will have to talk it over with my prof before
 upgrading to Asterisk 1.4. The productive system is currently running on
 1.2.6.
 Thanks
 Abhishek


  On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote:
  On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
   Dear Mr Gavin,
  
   Sorry for having miss pelt  your name twice... Thank you once again for
 your
   prompt reply. Is this the correct version of the driver for Asterisk
 1.2.x :
res_config_ldap-v0.7.tar.gz  from the link
   http://bugs.digium.com/view.php?id=5768
 
  If you use an old version of res_config_ldap with Asterisk version
  1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you
  seek any help via the lists or bug tracker.
 
  If you can use the latest release of Asterisk, you should.
 
  
   Thank you for your time and patience,
  
   Abhishek
  
  
  
  
On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
 Dear Mr Galvin,
   
Gavin! ;-)
   

 As of today I am using the res_config_ldap of Astirectory in my test
 Asterisk system to connect to a test LDAP database of my University.
   Things
 seem to be working fine so far. Now I'm faced with the task of
   installing
 this in the productive system. Before doing so, I'd sure like to
   consider
 trying the RealTime database driver that you people have developed.
 Why
   so?
 because I trust your judgment.
   
Thanks, but you should still test it yourself.
   

  I see it is res_config_ldap. You'd be much better using the
 latest
  version in the bug tracker.

 This would mean removing Astirectory module, installing the new
 driver
   and
 loading the new schema into LDAP. In my view, the latter part
 shouldn't
   be a
 concern because the old attributes and object classes (Astirectory)
   should
 in no way interfere with the new ones. Besides the old object
 classes
   could
 be deleted from LDAP. Also the former part shouldn't be of much
 concern
 either.
   
Nope, you are correct.
   

 My only concern as of now is in the installation of the RealTime
   database
 driver because the 'readme' file does not say anything about the
 installation. It only says about the configuration after
 installation.
 From the link:

  
 http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/
 Would it be sufficiant if I were to copy the makefile and
   res_config_ldap.c
 to the res/ directory of my running Asterisk and do make; make
 install?
   or
 do I have to do LIBS=-lldap export LIBS ./configure before that? My
   asterisk
 version is 1.2.6.
   
This Digium version is for 1.4.x, not 1.2
   

 Thanks in advance,
 Abhishek






 On 8/27/07, Gavin Henry  [EMAIL PROTECTED]  wrote:
  I see it is res_config_ldap. You'd be much better using the latest
  version in the bug tracker.
 
  On 27/08/07, Gavin Henry  [EMAIL PROTECTED] wrote:
   On 26/08/07, Abhishek M S  [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
  
   Gavin ;-)
  
   
Thank you for the links. Had gone through the bug tracker
 before
 though. I
was specifically referring to the schema for the driver
   'Astirectory'
 and
not the one related to the real time LDAP driver for Open
 LDAP.
  
   It's for any LDAP Compliant Directory Server.
  
In the
'Astirectory'  documentation there's a file defining the
 schema
   for
 LDAP
which is incomplete. By incomplete I mean the Syntax and few
 other
 fields
are not defined let alone the schema being a static file. I do
 understand
that for Open LDAP a static file schema should be defined.
  
   Not really. in the RealTime driver you can specify which LDAP
   attributes map to which Asterisk Config settings.
  
The only reason why I preferred Astirectory over the LDAP real
   time
 driver
was the fact that there is no mapping required for SIP users
 and
 peers.
  
   OK, maybe I need to go and read more about Astirectory.
  
   
Regards
Abhishek
   
   
On 8/24/07, Gavin Henry  [EMAIL PROTECTED] wrote:

 Please see the official tracker in the Digium buglist:

 http://bugs.digium.com/view.php?id=5768

 Here are the schemas we did for OpenLDAP:


   

  
 http://bugs.digium.com/file_download.php?file_id=14842type=bug

   

  
 http://bugs.digium.com/file_download.php?file_id=14841type=bug

 Also, for Novell eDirectory, see:



 http://forge.voicerd.org/frs/?group_id=7release_id=17

 Gavin

Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-28 Thread Gavin Henry

On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
 Dear Mr Gavin,

 Sorry for having miss pelt  your name twice... Thank you once again for your
 prompt reply. Is this the correct version of the driver for Asterisk 1.2.x :
  res_config_ldap-v0.7.tar.gz  from the link
 http://bugs.digium.com/view.php?id=5768

If you use an old version of res_config_ldap with Asterisk version
1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you
seek any help via the lists or bug tracker.

If you can use the latest release of Asterisk, you should.


 Thank you for your time and patience,

 Abhishek




  On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote:
  On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
   Dear Mr Galvin,
 
  Gavin! ;-)
 
  
   As of today I am using the res_config_ldap of Astirectory in my test
   Asterisk system to connect to a test LDAP database of my University.
 Things
   seem to be working fine so far. Now I'm faced with the task of
 installing
   this in the productive system. Before doing so, I'd sure like to
 consider
   trying the RealTime database driver that you people have developed. Why
 so?
   because I trust your judgment.
 
  Thanks, but you should still test it yourself.
 
  
I see it is res_config_ldap. You'd be much better using the latest
version in the bug tracker.
  
   This would mean removing Astirectory module, installing the new driver
 and
   loading the new schema into LDAP. In my view, the latter part shouldn't
 be a
   concern because the old attributes and object classes (Astirectory)
 should
   in no way interfere with the new ones. Besides the old object classes
 could
   be deleted from LDAP. Also the former part shouldn't be of much concern
   either.
 
  Nope, you are correct.
 
  
   My only concern as of now is in the installation of the RealTime
 database
   driver because the 'readme' file does not say anything about the
   installation. It only says about the configuration after installation.
   From the link:
  
 http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/
   Would it be sufficiant if I were to copy the makefile and
 res_config_ldap.c
   to the res/ directory of my running Asterisk and do make; make install?
 or
   do I have to do LIBS=-lldap export LIBS ./configure before that? My
 asterisk
   version is 1.2.6.
 
  This Digium version is for 1.4.x, not 1.2
 
  
   Thanks in advance,
   Abhishek
  
  
  
  
  
  
   On 8/27/07, Gavin Henry  [EMAIL PROTECTED]  wrote:
I see it is res_config_ldap. You'd be much better using the latest
version in the bug tracker.
   
On 27/08/07, Gavin Henry  [EMAIL PROTECTED] wrote:
 On 26/08/07, Abhishek M S  [EMAIL PROTECTED] wrote:
  Dear Mr Galvin,

 Gavin ;-)

 
  Thank you for the links. Had gone through the bug tracker before
   though. I
  was specifically referring to the schema for the driver
 'Astirectory'
   and
  not the one related to the real time LDAP driver for Open LDAP.

 It's for any LDAP Compliant Directory Server.

  In the
  'Astirectory'  documentation there's a file defining the schema
 for
   LDAP
  which is incomplete. By incomplete I mean the Syntax and few other
   fields
  are not defined let alone the schema being a static file. I do
   understand
  that for Open LDAP a static file schema should be defined.

 Not really. in the RealTime driver you can specify which LDAP
 attributes map to which Asterisk Config settings.

  The only reason why I preferred Astirectory over the LDAP real
 time
   driver
  was the fact that there is no mapping required for SIP users and
   peers.

 OK, maybe I need to go and read more about Astirectory.

 
  Regards
  Abhishek
 
 
  On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:
  
   Please see the official tracker in the Digium buglist:
  
   http://bugs.digium.com/view.php?id=5768
  
   Here are the schemas we did for OpenLDAP:
  
  
 
  
 http://bugs.digium.com/file_download.php?file_id=14842type=bug
  
 
  
 http://bugs.digium.com/file_download.php?file_id=14841type=bug
  
   Also, for Novell eDirectory, see:
  
  
   http://forge.voicerd.org/frs/?group_id=7release_id=17
  
   Gavin.
  
   --
  
 http://www.suretecsystems.com/services/openldap/
  
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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-27 Thread Gavin Henry

On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
 Dear Mr Galvin,

Gavin ;-)


 Thank you for the links. Had gone through the bug tracker before though. I
 was specifically referring to the schema for the driver 'Astirectory' and
 not the one related to the real time LDAP driver for Open LDAP.

It's for any LDAP Compliant Directory Server.

 In the
 'Astirectory'  documentation there's a file defining the schema for LDAP
 which is incomplete. By incomplete I mean the Syntax and few other fields
 are not defined let alone the schema being a static file. I do understand
 that for Open LDAP a static file schema should be defined.

Not really. in the RealTime driver you can specify which LDAP
attributes map to which Asterisk Config settings.

 The only reason why I preferred Astirectory over the LDAP real time driver
 was the fact that there is no mapping required for SIP users and peers.

OK, maybe I need to go and read more about Astirectory.


 Regards
 Abhishek


 On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:
 
  Please see the official tracker in the Digium buglist:
 
  http://bugs.digium.com/view.php?id=5768
 
  Here are the schemas we did for OpenLDAP:
 
 
 http://bugs.digium.com/file_download.php?file_id=14842type=bug
 
 http://bugs.digium.com/file_download.php?file_id=14841type=bug
 
  Also, for Novell eDirectory, see:
 
  http://forge.voicerd.org/frs/?group_id=7release_id=17
 
  Gavin.
 
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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-27 Thread Gavin Henry

I see it is res_config_ldap. You'd be much better using the latest
version in the bug tracker.

On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote:
 On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
  Dear Mr Galvin,

 Gavin ;-)

 
  Thank you for the links. Had gone through the bug tracker before though. I
  was specifically referring to the schema for the driver 'Astirectory' and
  not the one related to the real time LDAP driver for Open LDAP.

 It's for any LDAP Compliant Directory Server.

  In the
  'Astirectory'  documentation there's a file defining the schema for LDAP
  which is incomplete. By incomplete I mean the Syntax and few other fields
  are not defined let alone the schema being a static file. I do understand
  that for Open LDAP a static file schema should be defined.

 Not really. in the RealTime driver you can specify which LDAP
 attributes map to which Asterisk Config settings.

  The only reason why I preferred Astirectory over the LDAP real time driver
  was the fact that there is no mapping required for SIP users and peers.

 OK, maybe I need to go and read more about Astirectory.

 
  Regards
  Abhishek
 
 
  On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:
  
   Please see the official tracker in the Digium buglist:
  
   http://bugs.digium.com/view.php?id=5768
  
   Here are the schemas we did for OpenLDAP:
  
  
  http://bugs.digium.com/file_download.php?file_id=14842type=bug
  
  http://bugs.digium.com/file_download.php?file_id=14841type=bug
  
   Also, for Novell eDirectory, see:
  
   http://forge.voicerd.org/frs/?group_id=7release_id=17
  
   Gavin.
  
   --
   http://www.suretecsystems.com/services/openldap/
  
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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-27 Thread Gavin Henry

On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
 Dear Mr Galvin,

Gavin! ;-)


 As of today I am using the res_config_ldap of Astirectory in my test
 Asterisk system to connect to a test LDAP database of my University. Things
 seem to be working fine so far. Now I'm faced with the task of installing
 this in the productive system. Before doing so, I'd sure like to consider
 trying the RealTime database driver that you people have developed. Why so?
 because I trust your judgment.

Thanks, but you should still test it yourself.


  I see it is res_config_ldap. You'd be much better using the latest
  version in the bug tracker.

 This would mean removing Astirectory module, installing the new driver and
 loading the new schema into LDAP. In my view, the latter part shouldn't be a
 concern because the old attributes and object classes (Astirectory) should
 in no way interfere with the new ones. Besides the old object classes could
 be deleted from LDAP. Also the former part shouldn't be of much concern
 either.

Nope, you are correct.


 My only concern as of now is in the installation of the RealTime database
 driver because the 'readme' file does not say anything about the
 installation. It only says about the configuration after installation.
 From the link:
 http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/
 Would it be sufficiant if I were to copy the makefile and res_config_ldap.c
 to the res/ directory of my running Asterisk and do make; make install? or
 do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk
 version is 1.2.6.

This Digium version is for 1.4.x, not 1.2


 Thanks in advance,
 Abhishek






 On 8/27/07, Gavin Henry [EMAIL PROTECTED]  wrote:
  I see it is res_config_ldap. You'd be much better using the latest
  version in the bug tracker.
 
  On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote:
   On 26/08/07, Abhishek M S  [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
  
   Gavin ;-)
  
   
Thank you for the links. Had gone through the bug tracker before
 though. I
was specifically referring to the schema for the driver 'Astirectory'
 and
not the one related to the real time LDAP driver for Open LDAP.
  
   It's for any LDAP Compliant Directory Server.
  
In the
'Astirectory'  documentation there's a file defining the schema for
 LDAP
which is incomplete. By incomplete I mean the Syntax and few other
 fields
are not defined let alone the schema being a static file. I do
 understand
that for Open LDAP a static file schema should be defined.
  
   Not really. in the RealTime driver you can specify which LDAP
   attributes map to which Asterisk Config settings.
  
The only reason why I preferred Astirectory over the LDAP real time
 driver
was the fact that there is no mapping required for SIP users and
 peers.
  
   OK, maybe I need to go and read more about Astirectory.
  
   
Regards
Abhishek
   
   
On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:

 Please see the official tracker in the Digium buglist:

 http://bugs.digium.com/view.php?id=5768

 Here are the schemas we did for OpenLDAP:


   
 http://bugs.digium.com/file_download.php?file_id=14842type=bug

   
 http://bugs.digium.com/file_download.php?file_id=14841type=bug

 Also, for Novell eDirectory, see:


 http://forge.voicerd.org/frs/?group_id=7release_id=17

 Gavin.

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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file

2007-08-24 Thread Gavin Henry

Please see the official tracker in the Digium buglist:

http://bugs.digium.com/view.php?id=5768

Here are the schemas we did for OpenLDAP:

http://bugs.digium.com/file_download.php?file_id=14842type=bug
http://bugs.digium.com/file_download.php?file_id=14841type=bug

Also, for Novell eDirectory, see:

http://forge.voicerd.org/frs/?group_id=7release_id=17

Gavin.

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Re: [asterisk-users] ISDN30 card for UK : sanity check

2007-08-08 Thread Gavin Henry

Price. They are good cards, just bells and whistles plus the Echo
cancellation on the a101d. Ask Sangoma, their must have a reason for
still selling them ;-)

Gavin.

On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
Hi Gavin

Many thanks for the note. For what reason do you recommend the old a101
though?

Regards
Rory

On 07/08/07, Gavin Henry ([EMAIL PROTECTED]) wrote:
Very good. Sangoma cards are great. Get the a101d though. Nice wee review:

http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/

Voipon are great guys too. We resell for them.

On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
We will be connecting our Asterisk server to ISDN 30 and intend using
the Sangoma A101 card. The install location is in London (UK).

Sangoma card at Voipon
http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA

I would be grateful to hear if this is the right choice of card. Usage
reports would be helpful.
--
Rory Campbell-Lange
Director
Campbell-Lange Workshop Ltd.
[EMAIL PROTECTED]
www.campbell-lange.net

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Re: [asterisk-users] ISDN30 card for UK : sanity check

2007-08-07 Thread Gavin Henry

Very good. Sangoma cards are great. Get the a101d though. Nice wee review:

http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/

Voipon are great guys too. We resell for them.

On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
 We will be connecting our Asterisk server to ISDN 30 and intend using
 the Sangoma A101 card. The install location is in London (UK).

 Sangoma card at Voipon
 http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA

 I would be grateful to hear if this is the right choice of card. Usage
 reports would be helpful.

 Regards
 Rory

 --
 Rory Campbell-Lange
 Campbell-Lange Workshop Ltd.
 [EMAIL PROTECTED]
 www.campbell-lange.net

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Re: [asterisk-users] Phantom calls: Detecting hangup quicker

2007-06-07 Thread Gavin Henry


On 07/06/07, Stephen Bosch [EMAIL PROTECTED] wrote:

Gavin Henry wrote:
 Dear all,

 We seem to be getting phantom calls when a inbound caller via the
 legacy pbx hangups before
 the SIP handsets have answered. The extensions also seem to hear
 ringing on the lines too sometimes.

   SIP Inbound  
   |
 legacy pbx (analogue) - (sangoma a400d) asterisk - SIP phones

 Basically if a user hangups before the call has bridged, I think.

 Is there anything we can do about this?

Yet another call progress detection issue.


Ah, sorry. I didn't know the right terms to search for beforehand.



Analog lines are problematic this way. Search the archives for call
progress detection or disconnect supervision.


Many thanks.



-Stephen-
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[asterisk-users] Phantom calls: Detecting hangup quicker

2007-06-06 Thread Gavin Henry


Dear all,

We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes.

  SIP Inbound  
  |
legacy pbx (analogue) - (sangoma a400d) asterisk - SIP phones

Basically if a user hangups before the call has bridged, I think.

Is there anything we can do about this?

Thanks,

Gavin.
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Re: [asterisk-users] realtime ldap peer matching

2007-06-04 Thread Gavin Henry


On 04/06/07, Caio Zanolla [EMAIL PROTECTED] wrote:

Hi everyone,

in ldap realtime sip peers i need fullcontact set to
sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least
for the natted peers to reach them)...

anyway, how do I populate fullcontact on the fly with information from
exten and userip?


Wouldn't these just be dialplan vars?


of course, i could just do it staticaly on ldap but since the info is
already there why not make use of it?

on res_ldap.conf i have attribute = fullcontact = AstAccountFullContact
it would be nice to have something like:
attribute = fullcontact = sip:.$AstExten.@.$AstIPaddress
or some kind of dialplan scripting to archieve this...


I'm pretty sure res_ldap.c can't do this yet.

What version (* and res_ldap) and schema are you using btw?

IIRC, the latest version doesn't need:

attribute = fullcontact = AstAccountFullContact

just:

fullcontact = AstAccountFullContact


Thanks,

Gavin.




cheers,
Caio.


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[asterisk-users] ZAP inbound/outbound connection taking too long

2007-06-01 Thread Gavin Henry


Dear all,

I think this is common, or at least how it is supposed to be, but
whening dialing over a ZAP channel, it's taking around 5~ seconds to
ring on the over end, likewise inbound.

This is just with a normal Dial command.

Are there any ways to tweak this?

Thanks,

Gavin.
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Re: [asterisk-users] ZAP inbound/outbound connection taking too long

2007-06-01 Thread Gavin Henry


On 01/06/07, Gordon Henderson [EMAIL PROTECTED] wrote:

On Fri, 1 Jun 2007, Gavin Henry wrote:

 Dear all,

 I think this is common, or at least how it is supposed to be, but
 whening dialing over a ZAP channel, it's taking around 5~ seconds to
 ring on the over end, likewise inbound.

 This is just with a normal Dial command.

It's normal for an analogue Zap channel.

Asterisk has to sieze the line (after a basic check to make sure the
channel is free), that may entail a delay of a second or so while it makes
sure there there is a dial-tone (actually, I'm not sure it waits for a
dial-tone), then it sends the digits out via DTMF - that might take a
second or 2 for a long number - then it's up to the PSTN switch at the
other end to connect the call - depending on the technology, this might
take several seconds.

What you can do is connect to asterisk (asterisk -r), set verbose ,
then initiate a dial and you'll see the dialplan progress and you can work
out yourself where the longest part of the delay is...

Inbound ought to be answered as soon as asterisk hears the ringing
signal - but this might be one whole ring time from the ring starting,
depending on how caller-id is being handled in your country, again,
monitor it by looking at the output on the console, and by connecting an
existing analogue phone in paralel with the incoming Zap line.

Gordon
___



Thanks for this explaination!

Gavin.


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Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-30 Thread Gavin Henry


This is what is shown when the call connects with:

sip show channel

The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289

Internal between Asterisk and another Conference suite:

* SIP Call
Direction:  Outgoing
Call-ID:[EMAIL PROTECTED]
Our Codec Capability:   14
Non-Codec Capability:   1
Their Codec Capability:   4
Joint Codec Capability:   4
Format  ulaw
Theoretical Address:192.168.45.183:5605
Received Address:   192.168.45.183:2289
NAT Support:Always
Audio IP:   192.168.45.196 (local)
Our Tag:as31c610d6
Their Tag:  t1122b
SIP User agent:
Username:   slee
Peername:   slee
Original uri:   sip:[EMAIL PROTECTED]:5605
Need Destroy:   0
Last Message:   Tx: ACK
Promiscuous Redir:  No
Route:  sip:[EMAIL PROTECTED]:5605
DTMF Mode:  rfc2833
SIP Options:(none)

Inbound from SIP Provider:

* SIP Call
Direction:  Incoming
Call-ID:[EMAIL PROTECTED]
--   REMOVED
Our Codec Capability:   14
Non-Codec Capability:   1
Their Codec Capability:   14
Joint Codec Capability:   14
Format  gsm
Theoretical Address:193.111.201.32:5060
Received Address:   193.111.201.32:5060
NAT Support:Always
Audio IP:   xx.xx.xx.xx (local)
--   REMOVED
Our Tag:as65c31c43
Their Tag:  as26378dd7
SIP User agent: Asterisk PBX
Original uri:   sip:[EMAIL PROTECTED] --   REMOVED
Caller-ID:  01X
--   REMOVED
Need Destroy:   0
Last Message:   Rx: ACK
Promiscuous Redir:  No
Route:  sip:193.111.201.32;lr=on;ftag=as26378dd7
DTMF Mode:  rfc2833
SIP Options:(none)
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[asterisk-users] Theoretical and Received SIP addresses causing no audio

2007-05-29 Thread Gavin Henry


Hi,

This contacted call has no audio, any ideas?

The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289

Internal between Asterisk and another Conference suite:

 * SIP Call
 Direction:  Outgoing
 Call-ID:[EMAIL PROTECTED]
 Our Codec Capability:   14
 Non-Codec Capability:   1
 Their Codec Capability:   4
 Joint Codec Capability:   4
 Format  ulaw
 Theoretical Address:192.168.45.183:5605
 Received Address:   192.168.45.183:2289
 NAT Support:Always
 Audio IP:   192.168.45.196 (local)
 Our Tag:as31c610d6
 Their Tag:  t1122b
 SIP User agent:
 Username:   slee
 Peername:   slee
 Original uri:   sip:[EMAIL PROTECTED]:5605
 Need Destroy:   0
 Last Message:   Tx: ACK
 Promiscuous Redir:  No
 Route:  sip:[EMAIL PROTECTED]:5605
 DTMF Mode:  rfc2833
 SIP Options:(none)

Inbound from SIP Provider:

 * SIP Call
 Direction:  Incoming
 Call-ID:[EMAIL PROTECTED]
--   REMOVED
 Our Codec Capability:   14
 Non-Codec Capability:   1
 Their Codec Capability:   14
 Joint Codec Capability:   14
 Format  gsm
 Theoretical Address:193.111.201.32:5060
 Received Address:   193.111.201.32:5060
 NAT Support:Always
 Audio IP:   xx.xx.xx.xx (local)
 --   REMOVED
 Our Tag:as65c31c43
 Their Tag:  as26378dd7
 SIP User agent: Asterisk PBX
 Original uri:   sip:[EMAIL PROTECTED] --   REMOVED
 Caller-ID:  01X
--   REMOVED
 Need Destroy:   0
 Last Message:   Rx: ACK
 Promiscuous Redir:  No
 Route:  sip:193.111.201.32;lr=on;ftag=as26378dd7
 DTMF Mode:  rfc2833
 SIP Options:(none)
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Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-26 Thread Gavin Henry


On 23/05/07, Mojo with Horan  Company, LLC [EMAIL PROTECTED] wrote:

Does the non-Asterisk server _answer_ the line? :)


Hi, sorry. I have been away on site doing 8 work ;-)

Yes, it does. We've done a packet trace and it appears that * sends an
ACK back on the wrong port, i.e. not 5605 like a softphone does, in
the SDP session.



Gavin Henry wrote:
 Dear All,

 I have a tiny dial plan like:

 [testing]
 exten = 454,s,Ringing()
 exten = 454,n,Wait(4)
 exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
 exten = 454,n,Hangup


 This connects fine when I dial 454 from any extension in my system,
 but there is never any audio?

 Where can I start to look for debugging this? It's all internal so no
 NAT problems?

 Thanks,

 Gavin.
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Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-26 Thread Gavin Henry


On 23/05/07, Nick Seraphin [EMAIL PROTECTED] wrote:



The 2 most common problems I've seen for no audio in one or both
directions is usually either a firewall (which you already said you don't
have) or a CODEC problem.

Make sure both sides are negotiating the same CODEC.  I've often seen
situations where something like the Asterisk server will allow gsm, g711,
etc. and the phone is set for g711, but because gsm was first in the list
on the asterisk side, asterisk was trying to do gsm and the phone wanted
g711 and they wouldn't sync up.  It wasn't until I did a:

disallow=all
allow=g711

in sip.conf that it finally started working for me.

That may not be your exact problem, but my guess would be a CODEC issue if
it's not your firewall.


I'll check this out, thanks.



-- Nick


On Wed, 23 May 2007, Gavin Henry wrote:

 Dear All,

 I have a tiny dial plan like:

 [testing]
 exten = 454,s,Ringing()
 exten = 454,n,Wait(4)
 exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
 exten = 454,n,Hangup


 This connects fine when I dial 454 from any extension in my system,
 but there is never any audio?

 Where can I start to look for debugging this? It's all internal so no
 NAT problems?

 Thanks,

 Gavin.
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[asterisk-users] Voicemail and Time Conditions

2007-05-26 Thread Gavin Henry


Dear All,

With the standard Voicemail system, is it possible to have your
Busy/Unavailable messages only apply during say 9-5, then another
message saying you've gone home after that time?

It might be just a case of user training, that they change their
message if they need this feature.

A custom dialplan should be easy to do anyway, but I thought I'd ask.

Thanks,

Gavin.
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[asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Gavin Henry


Dear All,

I have a tiny dial plan like:

[testing]
exten = 454,s,Ringing()
exten = 454,n,Wait(4)
exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten = 454,n,Hangup


This connects fine when I dial 454 from any extension in my system,
but there is never any audio?

Where can I start to look for debugging this? It's all internal so no
NAT problems?

Thanks,

Gavin.
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Re: [asterisk-users] SIP Dial Command to a non-Asterisk url

2007-05-23 Thread Gavin Henry


On 23/05/07, Alex Balashov [EMAIL PROTECTED] wrote:


Gavin,


Hi.



   Does the Asterisk server's route to 192.168.45.183 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?


Nope, all internal.



   SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary
ports between the two endpoints for actually passing media.  If these are
being dropped somewhere along the way, you'll have no audio in one or
more directions of the call path.


Yeah, I understand that. It looks like * it not sending an ACK back to
the other SIP server, well it is, but not on the same port.



   Best thing to do is to is a packet capture on the Asterisk server and
filter on 192.168.45.183 to verify that you're seeing bidirectional media,
from and to that host.  Chances are something will be missing.


Yeah, we've done this, but it seems to be not replying to the correct port.



   Of course, it could be a non-IP problem of some sort as well, perhaps
even something fairly obvious.


Hmmm, hope so. This is the danger of too much knowledge.



-- Alex

--
Alex Balashov   [EMAIL PROTECTED]
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Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?

2007-05-10 Thread Gavin Henry


On 09/05/07, Cory Andrews [EMAIL PROTECTED] wrote:

Gavin - you should look at the Sangoma A4000X series cards, which only
occupy a single slot and come in PCI or PCI-X versions.


That was next on my list.

Thanks.




Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gavin
Henry
Sent: Wednesday, May 09, 2007 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 10 FXS - Channel Bank or PCI Card?

Hi All,

What do you recommend? I was looking at:

http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p
-393.html

But it will be 3 PCI slots.

Thanks,

Gavin.
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Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?

2007-05-10 Thread Gavin Henry


On 09/05/07, Robert Hajime Lanning [EMAIL PROTECTED] wrote:

I would look into one of these:
http://www.digium.com/en/products/hardware/analogcards.php


I've seen those too ;-)



quote who=Gavin Henry
 Hi All,

 What do you recommend? I was looking at:

 
http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html

 But it will be 3 PCI slots.

--
And, did Galoka think the Ulus were too ugly to save?
 -Centauri

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Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?

2007-05-10 Thread Gavin Henry


On 09/05/07, cb [EMAIL PROTECTED] wrote:

On May 9, 2007, at 3:45 PM, Gavin Henry wrote:

 http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-
 express-p-393.html

 But it will be 3 PCI slots.

Just to clarify in case you didn't already realize it. It doesn't
actually *use* 3 PCI slots, it just occupies the physical space of 3.
The board only connects to one slot, then has its own backplane that
the additional daughter cards sit on.

An important distinction if your concern with the use of 3 slots
wasn't due to physical space, but rather was with dealing with IRQ
and timing issues of having multiple slots in use.


That's a point, maybe if this card sits on the outer PCI slot there
would be enough space.

Hmmm...




-chris
www.mythtech.net


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