[asterisk-users] Contact Directory on Polycom phones
Hi, Polycom phones configured on asterisk pbx and are using contact directory on phones. To modify entries xml file for each phone needs to be modified and have to reboot all phones to accept updated file. Is there any way via asterisk, that we can use central database and on modification automatically update xml files on boot server and reboot phones. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact Directory on Polycom phones
I use the mini-web browser built into the phone and have a custom button (directory) that accesses the directory, which is hosted on a web server. It isn't perfect, but it's better than the XML files IMHO. That said, there's an enterprise license for these phones which enables directory integration. On Thu, Mar 3, 2011 at 9:20 AM, deeps backup backup.de...@gmail.com wrote: Hi, Polycom phones configured on asterisk pbx and are using contact directory on phones. To modify entries xml file for each phone needs to be modified and have to reboot all phones to accept updated file. Is there any way via asterisk, that we can use central database and on modification automatically update xml files on boot server and reboot phones. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
Your registration looks all wrong. The contact header appears
incorrect on this invite. Please make it read
Contact: sip:{broadsmart_us...@{our_ip}:5060
This is probably the userid or auth user id.
REGISTER sip:{broadsmart_ip} SIP/2.0
Via: SIP/2.0/UDP {our_ip}:5060;branch=z9hG4bK1e85dd83;rport
Max-Forwards: 70
From: sip:{broadsmart_us...@{broadsmart_ip};tag=as3bafb590
To: sip:{broadsmart_us...@{broadsmart_ip}
Call-ID: 13545ba119fb96b707e90636720df...@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Expires: 120
Contact: sip:s...@{our_ip}
Content-Length: 0
Please change expires to what we are configured which is 3600 seconds.
I'm not sure what it is that may be causing the Contact to show up as s.
Here are the associated configs.
sip.conf
[general]
register = {broadsmart_user}:{broadsmart_passwo...@{broadsmart_ip}
[broadsmart]
host={broadsmart_ip}
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser={broadsmart_user}
secret={broadsmart_password}
fromdomain=broadsmart.net
quality=3600
canreinvite=no
Sorry for the long request. Admittedly I'm lost.
--
Mike A. Leonetti
As warm as green tea
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
--
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
It turns out that it's actually on the registration end. I see that too:
[May 7 13:02:14] NOTICE[10402]: chan_sip.c:11461 sip_reregister:--
Re-registration for {broadsmart_passwo...@{broadsmart_ip}
REGISTER 12 headers, 0
lines
Reliably Transmitting (no NAT) to
{broadsmart_ip}:5060:
REGISTER sip:{broadsmart_ip}
SIP/2.0
Via: SIP/2.0/UDP
{asterisk_ip}:5060;branch=z9hG4bK6df043c0;rport
Max-Forwards:
70
From:
sip:{broadsmart_passwo...@{broadsmart_ip};tag=as59ede08c
To:
sip:{broadsmart_passwo...@{broadsmart_ip}
Call-ID:
4fd754b9115b2e1c2c17ce6d1f24b...@127.0.0.1
CSeq: 104
REGISTER
User-Agent: Asterisk PBX
1.6.2.5
Authorization: Digest username={broadsmart_password},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=c022714eff5d7016afe930e9390392a3,
response=2e14289556acb0bf2657504c9147b6c1,
opaque=e5677a6b
Expires:
3600
Contact:
sip:s...@{asterisk_ip}
Content-Length: 0
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Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
I see what it is. It was the contact extension value that wasn't set.
It defaults to s. Adding a / and putting that contact extension
afterwards fixed the problem. The phones still aren't working, but
thanks for all of the help.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
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[asterisk-users] Contact header gets url decoded?
I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inbound INVITE like: mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 The older version of asterisk replies with a 200 OK and a Contact: header that looks like: Contact: sip:mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 Newer 1.6 Asterisk (I've tried 1.6.0.9 and 1.6.0.27) take the identical call and reply with a 200 OK and a Contact header of: Contact: sip:mumble-fratz-sip:f...@bar.com@asteriskbox.abc.com:5062 And the calling applications appear to not recognize this 200 OK and never send an ACK and Asterisk eventually throws in the towel on the call setup Is there a knob I can adjust this behavior? The original To: is never molested in the same way, just the Contact header. Thanks in advance, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contact id protocol problem
Hi, I'm using an Asterisk box with zap channel as a gateway between PSTN and an alarm receiver system. The alarm system uses Contact ID protocol. My problem is that the negotiation fails and I think that the problem is that kissoff tone is cut and the transmitter doesn't recognize it. Maybe the asterisk tone duration isn't long enough. I'm thinking about increasing the toneduration value in zapata.conf. or changind DTMF tone frecuency. Does anyone deal with a similar problem? What are the optimal values? Thanks Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact lookup
On Thu, Feb 5, 2009 at 7:22 AM, Geoff Lane ge...@gjctech.co.uk wrote: The nice thing about that is that if I use MySQL I can run the management application on another machine, and so don't need to run a web server on the Asterisk box. However, I wonder whether the overhead necessary to run MySQL on the Asterisk box is more than that required to run Apache to provide a web interface to astdb. I'm not running either at present, which is probably as well since my Asterisk machine is low-spec by todays standards. Regarding system resource usage it is, of course, to you to run the DB engine along with asterisk or on some other system. :-) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact lookup
On Wednesday, February 4, 2009, D Tucny wrote:
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as yours above (including substitution of Unknown when not
found), but, all on one line...
Once I'd got a handle on it, the task seems almost trivial. Here's
what I've got:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I don't need unknown because all my handsets show something similar
(e.g. Unavailable name) by default.
I've been looking into changing it recently such that where I don't
have the name I can substitute something more useful than Unknown,
such as the site, or for external calls, the
country/province/state/city/type/telco/etc, though that won't be in
astdb due to the current 100s of thousands of rows...
That might be a good AGI project...
FWIW, I had trouble loading astdb with the contact list. I dumped the
caller list from my old PBX to a CSV file and then parsed it to give
one line of the following form for each contact:
/usr/sbin/asterisk -rx 'database put cidname 01234567 Caller Name'
I ran the script and it appeared to go well. However, when did
database show cidname at the * CLI prompt, the family and key were
in a right mess. For example, the entry above might have appeared as:
34567: Caller Name
I suspect that it would still have worked since database show
cidname listed these entries but I didn't take chances. There were
only thirty or so entries, so I cleared the cidname family and copied
each database put command from a terminal window and pasted to the *
CLI prompt.
So, this is now sorted for me and I've learned a thing or two about
astdb in the process.
Thanks all.
--
Geoff
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Re: [asterisk-users] Contact lookup
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
- Setup an ODBC connection for asterisk so that it can connect to that DB
(res_odbc.conf)
- Setup an ODBC func.This is basically an SQL query which will be
mapped into a dialplan function. (func_odbc.conf) It is essentially
something that states my function ODBC_LOOKUP(arg) will give me
the results of SELECT name FROM contactsTable WHERE number=${arg}
into the dialplan.
- Then use it in the dialplan
exten = _x.,n,Set(CALLERID(name)=${ODBC_LOOKUP(${EXTEN})})
There! Your dialplan is almost directly executing SQL queries. :)
Check both the sample asterisk configs + Asterisk TFOT, chapter 12.
It may be a bit more work than using the Ast DB or other means, but it
has the advantage of allowing the easy setup of any kind of frontend for
contact management.
Note: Check for the correctness of my filenames/syntax... They're shown
just to fill in the idea with something resembing the reality!
My 2c,
--
exvito
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Re: [asterisk-users] Contact lookup
On Wednesday, February 4, 2009, Ex Vito wrote: For a simple (but flexible) case I would consider ODBC + func_odbc. Here is the idea (in case you aren't aware of how it goes...) [... snip ...] It may be a bit more work than using the Ast DB or other means, but it has the advantage of allowing the easy setup of any kind of frontend for contact management. Thanks for the reply. The nice thing about that is that if I use MySQL I can run the management application on another machine, and so don't need to run a web server on the Asterisk box. However, I wonder whether the overhead necessary to run MySQL on the Asterisk box is more than that required to run Apache to provide a web interface to astdb. I'm not running either at present, which is probably as well since my Asterisk machine is low-spec by todays standards. At the moment it's academic since I don't have a large or extremely dynamic contact list and so can handle it with commands in the * CLI. However, it'll be an interesting exercise when I eventually upgrade the hardware and also move to Asterisk 1.6. Thanks again, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contact lookup
Hi All, Asterisk 1.4.12 on CentOS 5 I'd like to be able to look up each incoming CLI to retrieve an associated name, if available, and then pass that to the extensions so that they can see both the name and number of the caller. I'm not after LDAP or anything else maintained externally, just a contact lookup for my system. I suspect that Astdb could be used for this, as could a relational database like MySQL or postgres (accessed via AGI?) Probably simpler would be to maintain a text configuration file since I'm only concerned about less than a hundred entries initially. I'd appreciate insight into which is the easiest way to do this, and also any pointers to tutorials etc. TIA, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact lookup
There are some good examples of this at voip-info.org. Shouldn't this be
handled by normal caller-id?
Anyhow, here's an AGI (PERL) example:
#!/usr/local/bin/perl
use Asterisk::AGI;
# the AGI object
my $agi = new Asterisk::AGI;
# send callback reference
my $rc = $agi-set_callerid('IM_A_CALLER');
$agi-say_digits('123');
$agi-send_text('call ext 106');
$agi-exec('Dial', 'SIP/102');
$agi-hangup();
exit;
This sends the message call ext 106 to the phone display you call from,
says 123 on the speaker, then calls SIP/102 and shows IM_A_CALLER on the
display (the _ are there because IM A CALLER shows as IM.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Tuesday, February 03, 2009 10:05 AM
To: Asterisk Users
Subject: [asterisk-users] Contact lookup
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a contact
lookup for my system.
I suspect that Astdb could be used for this, as could a relational
database like MySQL or postgres (accessed via AGI?) Probably simpler
would be to maintain a text configuration file since I'm only
concerned about less than a hundred entries initially.
I'd appreciate insight into which is the easiest way to do this, and
also any pointers to tutorials etc.
TIA,
--
Geoff
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Re: [asterisk-users] Contact lookup
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a contact
lookup for my system.
I suspect that Astdb could be used for this, as could a relational
database like MySQL or postgres (accessed via AGI?) Probably simpler
would be to maintain a text configuration file since I'm only
concerned about less than a hundred entries initially.
I'd appreciate insight into which is the easiest way to do this, and
also any pointers to tutorials etc.
AstDB:
At it's very simplest:
exten = s,n,Set(CALLERID(name)=Unknown)
exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = s,n,GotoIf($[${name} = ]?endCID)
exten = s,n,Set(CALLERID(name)=${name})
exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
${CALLERID(all)})
... somewhere in the incoming processing. (This is an extract from an
overly complcated macro I use) Things to check for - a name already being
present - eg. on an incoming SIP call. No name in the astDB - might want
to substitute Unknown ..
All you need to do now is populate the astDB - I use a web interface and
some php to drive the manager interface...
My biggest site has just under 300 lookup entries... (Which presents other
issues with the web interface, but ...)
Gordon
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Re: [asterisk-users] Contact lookup
Have a look at smartCID at www.generationd.com
Uses a simple mySQL database, allows for call blocking flag, reverse CID
lookup, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: February 3, 2009 11:51 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Contact lookup
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a contact
lookup for my system.
I suspect that Astdb could be used for this, as could a relational
database like MySQL or postgres (accessed via AGI?) Probably simpler
would be to maintain a text configuration file since I'm only
concerned about less than a hundred entries initially.
I'd appreciate insight into which is the easiest way to do this, and
also any pointers to tutorials etc.
AstDB:
At it's very simplest:
exten = s,n,Set(CALLERID(name)=Unknown)
exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = s,n,GotoIf($[${name} = ]?endCID)
exten = s,n,Set(CALLERID(name)=${name})
exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
${CALLERID(all)})
... somewhere in the incoming processing. (This is an extract from an
overly complcated macro I use) Things to check for - a name already being
present - eg. on an incoming SIP call. No name in the astDB - might want
to substitute Unknown ..
All you need to do now is populate the astDB - I use a web interface and
some php to drive the manager interface...
My biggest site has just under 300 lookup entries... (Which presents other
issues with the web interface, but ...)
Gordon
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Re: [asterisk-users] Contact lookup
2009/2/4 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a contact
lookup for my system.
I suspect that Astdb could be used for this, as could a relational
database like MySQL or postgres (accessed via AGI?) Probably simpler
would be to maintain a text configuration file since I'm only
concerned about less than a hundred entries initially.
I'd appreciate insight into which is the easiest way to do this, and
also any pointers to tutorials etc.
AstDB:
At it's very simplest:
exten = s,n,Set(CALLERID(name)=Unknown)
exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = s,n,GotoIf($[${name} = ]?endCID)
exten = s,n,Set(CALLERID(name)=${name})
exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
${CALLERID(all)})
... somewhere in the incoming processing. (This is an extract from an
overly complcated macro I use) Things to check for - a name already being
present - eg. on an incoming SIP call. No name in the astDB - might want
to substitute Unknown ..
All you need to do now is populate the astDB - I use a web interface and
some php to drive the manager interface...
My biggest site has just under 300 lookup entries... (Which presents other
issues with the web interface, but ...)
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as yours above (including substitution of Unknown when
not found), but, all on one line...
I've been looking into changing it recently such that where I don't have the
name I can substitute something more useful than Unknown, such as the site,
or for external calls, the country/province/state/city/type/telco/etc,
though that won't be in astdb due to the current 100s of thousands of
rows...
d
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[asterisk-users] contact header is missing in 200OK for SUBSCRIBE
Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails, any body knows about this issue...? Regards, Subramanya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] contact header is missing in 200OK for SUBSCRIBE
Hi, Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails and also expires is 0 in the 200 OK any body knows about these issue...? On 8/23/07, sumanth achar [EMAIL PROTECTED] wrote: Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails, any body knows about this issue...? Regards, Subramanya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contact: header and NAT.
Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However, MetaSwitch doesn't like that; it expects (whether doing IP-trust or user authentication) to contact the SIP peer at such and such IP address in the SIP binding, and expects that's what the Contact: reachability information will be too. Any way to overcome this in Asterisk? I thought about the externip= option but it did not seem to work from an internal test box that is not behind NAT. Thanks, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact: header and NAT.
Got this figured out. externip= does work if the other NAT-related options are also enabled, plus it appears that Trixbox (which is what the end-user was using, it seems) handles this well in its config file structure regardless. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
Remarque : message transféré en pièce jointe. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be advice. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Contact Directory on Polycom IP-501 phones
On Thu, 2005-09-01 at 15:59 -0400, Jeremy Melanson wrote: Hi Jesse. A couple questions.. What firmware version are you using? Bootrom 2.6.2.20032 Sip 1.5.2.0054 How does your phone get it's config (FTP, TFTP, Manual config)? Initially it got the config from TFTP w/ the new boot rom. After that I did manual config on the phone. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact Directory on Polycom IP-501 phones
I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Contact Directory on Polycom IP-501 phones
Hi Jesse. A couple questions.. What firmware version are you using? How does your phone get it's config (FTP, TFTP, Manual config)? - Jeremy On Thu, 2005-09-01 at 12:51 -0700, Jesse Keating wrote: I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Contact Directory on Polycom IP-501 phones
On Thu, 2005-09-01 at 13:04 -0700, Jesse Keating wrote: Bootrom 2.6.2.20032 Sip 1.5.2.0054 I rolled back to Sip 1.4.1.0040 and I can save entries, but the menu system is all different and not easy to navigate. This is not so good. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact me Asap!
Hello Khurram, This is adnan from EBS kindly contact me as soon as possible i'll contact you on your number but its almost busy every time. Other *'s users kindly forgive me because i have no option right now. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] contact
Sorry to do this to the list, but I have no choice . Walker, I've been trying to send you an email off-list for the last couple of weeks, but one of my mail-hops is failing, do you have alternative address that I can try ??? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact header empty in SIP-message
Hi, I have noticed that when I am calling from my Snom-phone to another Snom-phone through Asterisk, the SIP-message's Contact -header could be sometimes empty and for example other Snom get no BYE-message. Here is example of that kind of message: 10 headers, 0 lines Sending to 192.168.0.32 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK-9sbwo79ob74p From: sip:[EMAIL PROTECTED];tag=wxrasulsor To: Mickey Mouse sip:[EMAIL PROTECTED];tag=as700f09c7 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Contact: Content-Length: 0 Same thing with INVITEs also... Does this error depend on my configuration or is it Asterisk's bug ? Asterisk built from CVS-07/11/03-14:12:25. Thank you for any help! -Johanna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
