Re: [asterisk-users] Default extension

[Olle E. Johansson]

On 26 Mar 2014, at 19:14, Mickael MONSIEUR  wrote:

> Hello,
> 
> When I get a SIP INVITE as follows: 
>> INVITE sip:s@10.1.0.191:5060 SIP/2.0
>> Max-Forwards: 69
>> From: "0475XX" ;tag=as7df9ab18
>> To: 
>> Contact: 
>> Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com
>> CSeq: 102 INVITE
>> Date: Wed, 26 Mar 2014 15:06:01 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
>> PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 252
>> 
> 
> Asterisk considers that the extension is 's'. (The Register) 
> How to make the extension number that is shown in the 'To' ??

You never route calls on the To: header in SIP. You route on the request URI. 
Unless this is something where you used the REGISTER statement in sip.conf and 
forgot to add an extension or you register once for multiple DIDs.

I would suggest changing your register statement to include an extension. In 
that extension you read the To: header with the SIP_HEADER() dialplan function 
and issue a goto so you end up with the extension in the To header.

The IETF has with help of the SIP forum written a standard extension to SIP to 
handle this use-case, something called GIN. It's now part of the SIPConnect 
specification. using the gin extension, you would get the called phone number 
in the r-uri.

/O

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Re: [asterisk-users] realtime sip.conf and templates

[Olle E. Johansson]

6 jun 2013 kl. 17:41 skrev Daniel Pocock :

> On 06/06/13 15:51, Daniel Pocock wrote:
>> Is the template capability in sip.conf compatible with realtime sip.conf
>> entries such as users in a database?
>> 
>> I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
>> don't mention a template column:
>> 
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>> 
>> while some third-party examples do suggest that a column named
>> "template" is permitted:
>> 
>> http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
> 
> I have actually tried adding that column "template" into sippeers and
> setting the value as the name of a template from my sip.conf - on
> Asterisk 11.4, it seems to ignore the column.  If there is a way to do
> this, it would be useful to have it in the wiki.
> 
The templates are part of the configuration file (text files) parser and 
not supported in databases.

/O

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Re: [asterisk-users] IPv6 and IPv4 binding address on a server with 2 network cards

[Olle E. Johansson]

13 mar 2013 kl. 20:30 skrev Miguel Baptista :

> Thank you Olle.
> 
> Well, in my test scenario I will leave the bindaddr=:: value. 
> Any idea if/when asterisk will support two specific bindaddr (one for IPv4 
> and another for IPv6) ?

I do hope that the new stack that will be in early beta in Asterisk 12 is going 
to handle that. To fix it for
current asteirsk would be messy, so I just have the habit of running Kamailio 
in front to handle
both these issues, as well as to add DoS protection and proper TCP/TLS support.

Like many other things, these kind of issues will be fixed when there's someone 
that needs to 
fix it and provides funding for a developer to do it or have developer 
resources, fix it and contribute
the code back to the project. 

/O
> 
> - Miguel Baptista
> 
> 
> On 3/13/2013 10:06 AM, Olle E. Johansson wrote:
>> 
>> ; With the current situation, you can do one of four things:
>> ;  a) Listen on a specific IPv4 address.  Example: bindaddr=192.0.2.1
>> ;  b) Listen on a specific IPv6 address.  Example: bindaddr=2001:db8::1
>> ;  c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0
>> ;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
>> 
>> 
>> From the configuration file - you can do "ONE" of four things, but only one. 
>> You can not have multiple bindaddr= settings.
>> 
>> /O
>> 11 mar 2013 kl. 11:03 skrev Miguel Baptista :
>> 
>>> Hi Asghar,
>>> 
>>> Thanks for you reply. Which Asterisk version are you using? 
>>> 
>>> I am using Asterisk 11.1.0 
>>> when I use the bindaddr  parameters with specific IP addresses, Asterisk 
>>> will listen only on the last entry. 
>>> 
>>> For example, when I have 
>>> bindaddr=ipv4A:port 
>>> bindaddr=[ipv6A]:port 
>>> 
>>> it will listen only on the IPv6A address
>>> 
>>> and when I have the other way around:
>>> 
>>> bindaddr=[ipv6A]:port 
>>> bindaddr=ipv4A:port 
>>> 
>>> Asterisk will only listen on the IPv4A address.
>>> 
>>> The only way I found to force asterisk to listen on both IPv4A and IPv6 A 
>>> was to use bindaddr=[::] but it makes asterisk to listen also on the other 
>>> IP addresses.
>>> 
>>> Maybe this is fix on a newer Asterisk version.
>>> 
>>> - Miguel Baptista
>>> 
>>> On 3/10/2013 8:04 PM, Asghar Mohammad wrote:
>>>> 
>>>> hi,
>>>> i am using similer setup just put  bindaddr=ipv4A:port and 
>>>> bindaddr=[ipv6A]:port ans it should work.
>>>> 
>>>> On Sun, Mar 10, 2013 at 3:04 PM, Miguel Baptista 
>>>>  wrote:
>>>> Hello,
>>>> 
>>>> I am doing some tests with asterisk on a dual-stack environment.  I have 
>>>> some doubts regarding asterisk binding addresses on a server with 2 
>>>> network cards.
>>>> 
>>>> According to asterisk documentation:
>>>> ; With the current situation, you can do one of four things:
>>>> ;  a) Listen on a specific IPv4 address.  Example: bindaddr=192.0.2.1
>>>> ;  b) Listen on a specific IPv6 address.  Example: bindaddr=2001:db8::1
>>>> ;  c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0
>>>> ;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
>>>> ; (You can choose independently for UDP, TCP, and TLS, by specifying 
>>>> different values for
>>>> ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
>>>> ; (Note that using bindaddr=:: will show only a single IPv6 socket in 
>>>> netstat.
>>>> ;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
>>>> ;
>>>> ; You may optionally add a port number. (The default is port 5060 for UDP 
>>>> and TCP, 5061
>>>> ; for TLS).
>>>> ;   IPv4 example: bindaddr=0.0.0.0:5062
>>>> ;   IPv6 example: bindaddr=[::]:5062
>>>> ;
>>>> ; The address family of the bound UDP address is used to determine how 
>>>> Asterisk performs
>>>> ; DNS lookups. In cases a) and c) above, only A records are considered. In 
>>>> case b), only
>>>> ;  records are considered. In case d), both A and  records are 
>>>> considered. Note,
>>>> ; however, that Asterisk ignores all records except the first one. In case 
>>>> d), when both A
>>>> ; and  records are availa

Re: [asterisk-users] How does Asterisk handle ACK's?

[Olle E. Johansson]

12 mar 2013 kl. 16:54 skrev Grant Bagdasarian :

> Hello,
>  
> I’m noticing strange behavior in one of our Asterisk nodes where the ACK is 
> always sent to the proxy, but RR is not enabled for calls.
> The proxy drops the ACK.
>  
> I’m using the AMI interface to originate a call:
> Action: login
> Username: myusername
> Secret: mypassword
> Events: on
>  
> Action: Originate
> Channel: SIP/@proxy1
> CallerID: 
> Application: Playback
> Data: hello-world
>  
> Using Asterisk 10.5.0.
>  
> Shouldn’t Asterisk send the ACK directly to the endpoint in the Contact 
> header?
Yes it should. File a bug report with details.

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Re: [asterisk-users] IPv6 and IPv4 binding address on a server with 2 network cards

[Olle E. Johansson]

; With the current situation, you can do one of four things:
;  a) Listen on a specific IPv4 address.  Example: bindaddr=192.0.2.1
;  b) Listen on a specific IPv6 address.  Example: bindaddr=2001:db8::1
;  c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0
;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::


From the configuration file - you can do "ONE" of four things, but only one. 
You can not have multiple bindaddr= settings.

/O
11 mar 2013 kl. 11:03 skrev Miguel Baptista :

> Hi Asghar,
> 
> Thanks for you reply. Which Asterisk version are you using? 
> 
> I am using Asterisk 11.1.0 
> when I use the bindaddr  parameters with specific IP addresses, Asterisk will 
> listen only on the last entry. 
> 
> For example, when I have 
> bindaddr=ipv4A:port 
> bindaddr=[ipv6A]:port 
> 
> it will listen only on the IPv6A address
> 
> and when I have the other way around:
> 
> bindaddr=[ipv6A]:port 
> bindaddr=ipv4A:port 
> 
> Asterisk will only listen on the IPv4A address.
> 
> The only way I found to force asterisk to listen on both IPv4A and IPv6 A was 
> to use bindaddr=[::] but it makes asterisk to listen also on the other IP 
> addresses.
> 
> Maybe this is fix on a newer Asterisk version.
> 
> - Miguel Baptista
> 
> On 3/10/2013 8:04 PM, Asghar Mohammad wrote:
>> 
>> hi,
>> i am using similer setup just put  bindaddr=ipv4A:port and 
>> bindaddr=[ipv6A]:port ans it should work.
>> 
>> On Sun, Mar 10, 2013 at 3:04 PM, Miguel Baptista 
>>  wrote:
>> Hello,
>> 
>> I am doing some tests with asterisk on a dual-stack environment.  I have 
>> some doubts regarding asterisk binding addresses on a server with 2 network 
>> cards.
>> 
>> According to asterisk documentation:
>> ; With the current situation, you can do one of four things:
>> ;  a) Listen on a specific IPv4 address.  Example: bindaddr=192.0.2.1
>> ;  b) Listen on a specific IPv6 address.  Example: bindaddr=2001:db8::1
>> ;  c) Listen on the IPv4 wildcard.Example: bindaddr=0.0.0.0
>> ;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
>> ; (You can choose independently for UDP, TCP, and TLS, by specifying 
>> different values for
>> ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
>> ; (Note that using bindaddr=:: will show only a single IPv6 socket in 
>> netstat.
>> ;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
>> ;
>> ; You may optionally add a port number. (The default is port 5060 for UDP 
>> and TCP, 5061
>> ; for TLS).
>> ;   IPv4 example: bindaddr=0.0.0.0:5062
>> ;   IPv6 example: bindaddr=[::]:5062
>> ;
>> ; The address family of the bound UDP address is used to determine how 
>> Asterisk performs
>> ; DNS lookups. In cases a) and c) above, only A records are considered. In 
>> case b), only
>> ;  records are considered. In case d), both A and  records are 
>> considered. Note,
>> ; however, that Asterisk ignores all records except the first one. In case 
>> d), when both A
>> ; and  records are available, either an A or  record will be first, 
>> and which one
>> ; depends on the operating system. On systems using glibc,  records are 
>> given
>> ; priority.
>> 
>> Lets say that I have two network cards: A and B. 
>> Both interfaces have IPv4 and IPv6 addresses: IPv4 A, IPv6 A, IPv4 B and 
>> IPv6 B.
>> 
>> How can I make asterisk to run only on B network addresses (IPv6 and IPv4)? 
>> The bindaddr=[::] config parameter tells asterisk to run on all available 
>> addresses, including the addresses on the A network. But that's not exactly 
>> what I want.
>> 
>> - Miguel Baptista
>> 
>> 
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>> 
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Re: [asterisk-users] Register Free Opensips/Asterisk Integration

[Olle E. Johansson]

10 mar 2013 kl. 03:04 skrev Nick Khamis :

> Hello Everyone,
> 
> I have gone through a few really good tutorials from the OpenSIPS
> site, Asterisk resources etc.. The unanswered question (and final
> piece of our puzzle) is if it's possible to have a register free
> environment in an OpenSIPS/Asterisk integration. Most approaches have
> OpenSIPS relay the UA's REGISTER request to Asterisk which has
> "host=dynamic" set for the Friend/Peer and everything works as
> expected.
> 
There are a lot of models for this. Check my presentation from Astricon
2010 to get some ideas.
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations

/O
> Where I run into problems is in Inbound calls. When I try to call the
> extension from a DID I am receiving "Unable to create channel of type
> 'SIP' (cause 20 - Unknown)". And rightfully so!
> Reason being:
> 
> SIP Show Peers Yields:
> 
> Name/username HostDynForcerport ACL Port
> Status   Realtime
> 1001/1001  192.168.2.5N  5060
> UNREACHABLE Cached RT
> TTrunk/sip.exp.com 192.168.2.5N  5060 UNKNOWN Cached RT
> 
> 
> As for who will keep track of the UA location, the OpenSIPS `location`
> table has the correct
> info:
> 
> select username,domain,contact,socket from location;
> +--+++--+
> | username | domain | contact| socket
> |
> +--+++--+
> | 1001 | sip.exp.com | sip:1001@192.168.2.11:5060 | udp:192.168.2.5:5060 |
> +--+++--+
> 
> OpenSIPS: sip.exp.com
> OpenSIPS: 192.168.2.5
> Asterisk: 192.168.2.10
> UA: 192.168.2.11
> 
> I have set `host=sip.exp.com' for the UA but the UA is still
> `UNREACHABLE` by asterisk
> 
> As for the rest of the media related stuff, everything works
> perfectly. Outbound works fine. As you know, this only poses a problem
> with inbound calls to the UAs.
> 
> Your Help is Greatly Appreciated,
> 
> Nick.
> 
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Re: [asterisk-users] BLF and call-limit in 1.8

[Olle E. Johansson]

6 dec 2012 kl. 16:54 skrev Danny Nicholas :

> Not sure about this since I use the 10/11 branches and not 1.8, but I think 
> you need to use the deprecated call-limit for BLF and the new busylimit for 
> the other features you need.
> http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf


Call-limit is the limit on the number of calls you can take and also sets a 
device to BUSY. Since you want to be able to transfer calls, you need at least 
two. But this did not set the phone to busy on one call. That's why we added 
busy-limit that can be set to the level you want device states to signal busy, 
but still give the ability to the phone to set up more calls.

counteronpeer is the same as limitonpeer, just a new name.

/O
>  
>  
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. 
> Christensen
> Sent: Thursday, December 06, 2012 9:50 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] BLF and call-limit in 1.8
>  
> Hello
>  
> We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF 
> lamps on our Polycom phones stop working. After a lot of googling and a lot 
> of testing, I have been unable to find a solution.
>  
> I did try to change the call-limit value from 4 to 1, and this actually made 
> BLF work (noone suggested this, and what documantation I can find states that 
> this option is deprecated). This change has other implications, however. Call 
> waiting stops working, queues don't offer calls if the user is in a private 
> call etc.
>  
> We have customers that require both BLF and call waiting at the same time.
>  
>  
> We are running Asterisk 1.8.11-cert7
>  
> I've made the following additions to sip.conf [general]:
> callcounter=yes
> counteronpeer=yes (undocumented? Supposed to replace limitonpeers?)
>  
> (old relevant values, unchanged)
> allowsubscribe=yes
> subscribecontext=blf
> notifyringing=yes
> notifyhold=yes
> limitonpeers=yes 
>  
> I also tried may other suggestions I've found like placing the hints in the 
> same context as the extensions and removing subscribecontext.
>  
> Is there something I'm missing? Is something not working correctly?
>  
> Thanks in advance,
> Pan
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Re: [asterisk-users] SIP - Authenticated vs Unauthenticated Calls

[Olle E. Johansson]

1 nov 2012 kl. 15:13 skrev Joshua Colp :

> Tim Nelson wrote:
>> 
>> Thanks Joshua-
>> 
>> In this case, we're using SIP registration to peer the remote systems to the 
>> 'central system'. In option #1 above, the 'user' portion is always the CID 
>> we set for the outbound call, but the actual SIP user is something different 
>> like 'site12' for example. So, it would appear #1 is not a match...
> 
> Registration just tells the remote system what your IP address/port is for 
> sending calls.
> 
> You don't *have* to send CID like you are. You can override using fromuser to 
> ensure that the specific peer is *always* matched and authenticated. CID can 
> be conveyed in an alternate header, like Remote-Party-ID. The options 
> involved for RPID are "sendrpid=yes" on the caller box and "trustrpid=yes" on 
> the receiving box.
> 
>> That leaves us with option #2. We're using 'qualify=yes' on both sides of 
>> the SIP peering. If a peer becomes unreachable (fast UDP state table timeout 
>> on a remote firewall for example) as seen by the central system, and an 
>> outbound call is made from the remote system, that would mean the call is 
>> coming from an unknown IP:port. Would this then make sense Asterisk would 
>> simply throw it into the from-sip-external context as an 
>> unknown/unauthenticated call? And of course, when the peer *is* registered, 
>> and a call is made, Asterisk on the central system allows the call as 
>> authenticated due to the source IP/port being known via the registration 
>> status?
> 
> It's possible, without logs and such it's only a guess.
Agree, all comments are pure speculations at this point.

Try removing the user object to simplify. If you have type=friend, change to 
type=peer and you will *only* get IP/port-based matching and can configure your 
system in a controlled way. There are just a few situations where you actually 
benefit from having type=friend and match object names with Caller ID numbers.

/O


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Re: [asterisk-users] Asterisk and OpenLDAP

[Olle E. Johansson]

31 okt 2012 kl. 15:07 skrev Giuseppe Longo :

> I don't want update Asterisk configuration, i want to query LDAP only
> for name and secret field.
> 
Currently Asterisk can't do that. If you add Kamailio as a proxy in front of 
Asterisk, you can
easily authenticate with LDAP this way. There was some work by Philippe Sultan 
in this area
done years ago, but was never completed.

In SIP, the MD5 Digest authentication is based on the cleartext password being 
available
to calculate the hash. Therefore we can't use the LDAP authentication for 
binding as an authentication
mechanism in SIP. As long as we can have a binding (authentication for the 
server itself)
and query and in the result get a cleartext authentication username and secret,
kamailio should be able to do the job.

The Asterisk realtime driver assumes that you use a [peer] or [user] object 
like the ones
we use in a database - or that you query from the dialplan with the realtime 
function.
However, as stated earlier, this doesn't work in the SIP authentication that is 
based on
the data in peers and users.

Regards,
/Olle



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Re: [asterisk-users] Why all the 401 Unauthorized

[Olle E. Johansson]

23 okt 2012 kl. 14:28 skrev Steven Howes :

> Hi,
> 
> SIP registrations typically try to register, are them prompted for a password 
> (via a 401 message) it then sends a new request with authentication . This is 
> normal.
> 
Yes it is.

To be more exact and clarify: We never prompt for a password or ask anyone to 
send a password in clear text. SIP authentication
is based on HTTP MD5 digest authentication, where we have to send an 
authentication challenge to get an authentication response.
The challenge is sent in the first 401 and the response in the next SIP request 
- REGISTER, INVITE or something else - from
the client. The information sent over the net is just a digest of a set of 
information, where the shared secret is one piece.

Cheers,
/O


> Steve
> 
> On 23 Oct 2012, at 13:26, Jerry Geis wrote:
> 
>> I have a connection between two asterisk boxes, both running 1.4.43
>> 
>> The connection is alive and good and working. however, I see a bunch of
>> 401 Unauthorized messages using wireshark - then it eventually registers 
>> again
>> just fine.
>> 
>> Why would it not successfully register the first time - every time?
>> 
>> Jerry
>> 
>> 

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Re: [asterisk-users] Wireshark AMI Dissector

[Olle E. Johansson]

23 okt 2012 kl. 22:31 skrev Kristian Kielhofner :

> Hello everyone,
> 
>  Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector?
> 
>  Decode as telnet and display filter telnet.data kind of work but TCP
> reassembly can't happen without a better understanding of the
> protocol...
> 
No, but that's a very cool idea. Would be great to have.

Cheers,
/O


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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Olle E. Johansson]

31 aug 2012 kl. 16:58 skrev Shitian Long :

> Do you think it is a good way to use Manager API "command action" to 
> implement this feature?
No. The command action should be avoided since the output from the CLI commands 
is not
made for parsing by applications and may change too. Sometimes we cut of 
informaiton to fit
into a terminal window. If you use manager actions instead, you will always get 
the full data in a
format you can parse. If you have to use the command action you have found a 
place where a
manager action is missing and we developers would like to know that and fix it 
:-)

For realtime, there's a dialplan function REALTIME() that you can use with the 
manager actions
that change or read channel variables. That's the best way, since we lack 
manager realtime commands.

One reason for not going directly to the database API is that when building 3rd 
party apps,
we don't know what database you are using and can benefit from the ARA 
interface to databases,
exactly like Asterisk. It's not as effecient as going directly when you can, 
but sometimes you just
don't know what's behind ARA and thanks to ARA you don't have to. :-)

/Olle

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Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

[Olle E. Johansson]

31 aug 2012 kl. 13:13 skrev Steve Davies :

> On 31 August 2012 07:49, Olle E. Johansson  wrote:
>> 
>> 24 aug 2012 kl. 16:18 skrev Steve Davies :
>> 
>>> Hi SIP Gurus,
>>> 
>>> I've tried to find the relevant RFCs, but am struggling. I can find
>>> the odd opinion online, but was wondering if anyone could give a
>>> definitive answer.
>>> 
>>> If a SIP call is initiated (INVITE) and receives either a "180 with
>>> SDP", or a "183 with SDP", then the remote party will start to send
>>> audio for the inband-ringing. Asterisk then passes this audio, and it
>>> is correctly heard by the caller.
>>> 
>>> At present, Asterisk will also start to pass back any handset audio in
>>> return, in theory allowing a conversation to occur on an unanswered
>>> channel if an endpoint were designed to allow this (free phonecalls
>>> here we come!).
>>> 
>>> My question:
>>> 
>>> Should:
>>> 1) Asterisk block outbound audio between the 183 Progress and the 200
>>> OK packets?
>>> 2) Replace any outbound audio with silence?
>>> 3) Replace outbound audio with a special NULL RTP of some sort? Does that 
>>> exist?
>>> 4) Allow any audio to be sent regardless?
>>> 
>>> I have implemented 1) at present on our test rig, but the lack of
>>> outbound RTP causes issues with firewall state not being set-up to
>>> allow the inbound audio. I am not sure how hard/easy it would be to do
>>> 2) as you'd need to create silence of the correct duration to replace
>>> each audio frame.
>>> 
>>> Thoughts please?
>> 
>> First, because of Asterisk's RTP implementation we have to send some RTP 
>> packets at this point. You could mute the calling channel before calling and 
>> unmute the channel at answer if needed, but normally sending audio won't 
>> hurt. A normal user should not be able to send early media on a pstn-like 
>> installation where you bill the calls, so there should be little risc of 
>> two-way conversations before an answer.
>> 
>> In some cases you have to let the caller send DTMF (the famous fed ex 
>> example) in
>> early media, so we can't block any media by default in Asterisk.
>> 
>> Using the "r" option in dial causes a lot of issues, since you can still get 
>> busy or congestion when you have early media, so that is not a good solution.
>> 
>> /Olle
>> 
> 
> Excellent information as always Olle. Many thanks.
> 
> My intention is to make the early-audio prevention in SIP a little
> more harsh, such that if SIP receives audio before a 183 or 200 is
> received, it is dropped.
> 
> This fixes the case where "useless" early-audio is received from a
> non-SIP (eg ISDN) technology, and can cause an onward node to
> auto-enable early audio mode, causing silent ringing and other broken
> behaviours.

This is one of my pet issues. THe problem today is that many gateway vendors 
ALWAYS send 183 with sdp,
regardless if it's a ring tone or a service provider message. If you kill the 
183, service provider messages
will disappear. My recommendation (which I've mentioned in tons of mails and 
blog entries) is to send
180 ringing with SDP for ring tones and 183 for other messages. That way I 
could kill the 180 SDP in a Kamailio
proxy before it hits the Asterisk server. In reality today, by killing 183 you 
will also block important information
for the caller, like "this subscriber has a new number".

/O

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Re: [asterisk-users] change channel variable to a user chosen value during a call

[Olle E. Johansson]

31 aug 2012 kl. 09:18 skrev Frederic Van Espen :

> On Fri, 2012-08-31 at 00:11 +, Andrew White wrote:
>> Is realtime an option for you to install?
> 
> Andrew,
> 
> Realtime is not an option actually. We have a whole system built up that
> generates configuration files.
> 
> The primary goal is to let the user directly change the channel variable
> with his phone, while in conversation, or with a short interruption of
> the call.
> 
> If that isn't possible, an AMI call will be just fine. I'd just like to
> make sure it is not possible on the phone itself first.

There is a "hidden" feature for SNOM phones in the SIP channel. They have a way 
to send a client
code during the call (made for lawyers) and that will end up in the CDR.

/O

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Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

[Olle E. Johansson]

24 aug 2012 kl. 16:18 skrev Steve Davies :

> Hi SIP Gurus,
> 
> I've tried to find the relevant RFCs, but am struggling. I can find
> the odd opinion online, but was wondering if anyone could give a
> definitive answer.
> 
> If a SIP call is initiated (INVITE) and receives either a "180 with
> SDP", or a "183 with SDP", then the remote party will start to send
> audio for the inband-ringing. Asterisk then passes this audio, and it
> is correctly heard by the caller.
> 
> At present, Asterisk will also start to pass back any handset audio in
> return, in theory allowing a conversation to occur on an unanswered
> channel if an endpoint were designed to allow this (free phonecalls
> here we come!).
> 
> My question:
> 
> Should:
> 1) Asterisk block outbound audio between the 183 Progress and the 200
> OK packets?
> 2) Replace any outbound audio with silence?
> 3) Replace outbound audio with a special NULL RTP of some sort? Does that 
> exist?
> 4) Allow any audio to be sent regardless?
> 
> I have implemented 1) at present on our test rig, but the lack of
> outbound RTP causes issues with firewall state not being set-up to
> allow the inbound audio. I am not sure how hard/easy it would be to do
> 2) as you'd need to create silence of the correct duration to replace
> each audio frame.
> 
> Thoughts please?

First, because of Asterisk's RTP implementation we have to send some RTP 
packets at this point. You could mute the calling channel before calling and 
unmute the channel at answer if needed, but normally sending audio won't hurt. 
A normal user should not be able to send early media on a pstn-like 
installation where you bill the calls, so there should be little risc of 
two-way conversations before an answer.

In some cases you have to let the caller send DTMF (the famous fed ex example) 
in 
early media, so we can't block any media by default in Asterisk.

Using the "r" option in dial causes a lot of issues, since you can still get 
busy or congestion when you have early media, so that is not a good solution.

/Olle

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Re: [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device

[Olle E. Johansson]

17 aug 2012 kl. 03:15 skrev Phillip Frost:

> 
> On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote:
> 
>> forward to a Local extension that has dialplan requiring the 
>> acknowledgement? 
>> 
>> On 16 August 2012 21:12, Phil Frost  wrote:
>> I'd like to allow my users to forward their calls using the forwarding 
>> feature on their SIP handsets and continue to receive Queue() calls. 
>> Currently I set the 'i' option in Queue() so that if a user forwards to 
>> their cell phone, or any other extension that has voicemail, the voicemail 
>> doesn't eat all the calls to the queue.
> 
> I'd think that would require teaching all the users to forward to a different 
> extension if they thought they could be receiving queue calls. My users 
> probably aren't that good at following directions ;)
> 
> Ultimately, I'm sure I could solve this problem by taking management of 
> forwarding off the phone and into Asterisk, since then I'd absolutely have 
> some flag indicating if forwarding is active or not. However, I was just 
> hoping there was an easier way. I'm really happy with the forwarding 
> interface on our current handsets, and I'd rather not go through the effort 
> of changing their configuration, or changing the user experience if I can 
> avoid it.

If a call is forwarded and hit the dialplan again, it's forwarded to the 
context set in the channel variable FORWARD_CONTEXT.

So you could set this variable before you hit queue(), then do things 
differently in the context specified by this variable, since you know that the 
call is forwarded.

/O
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Re: [asterisk-users] RFC List

[Olle E. Johansson]

8 aug 2012 kl. 14:07 skrev Kevin P. Fleming:

> On 08/08/2012 06:30 AM, Kannan wrote:
> 
>> Where can I get a complete set of RFCs and other specifications
>> supported by Asterisk?
> 
> To my knowledge there is no such list. In addition, Asterisk (like many other 
> pieces of software) does not claim 100% compliance with every RFC that is 
> relevant, so usually it's better to ask about the specific features you are 
> interested in.

THis is a document I haven't updated since 1.6.x but still covers a large part 
of the SIP implementation:
http://svnview.digium.com/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup


/O
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Re: [asterisk-users] sip.conf and binaddr issue

[Olle E. Johansson]

10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:

> On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
> 
>> The Asterisk SIP channel has no knowledge about interfaces and can't
>> bind to a specific interface for communication. In fact, it's a well known
>> bug that if you have multiple interfaces with different IP networks,
>> Asterisk will send from the wrong IP on some of the interfaces.
> 
> Are you sure about that? The only problem area that I'm aware of is when 
> there are multiple *overlapping* interfaces (on the same subnet, or providing 
> the same route(s)). In that case, Asterisk can receive messages on one IP 
> address out of the overlapping set, but reply using a different one from the 
> set, because it doesn't specify the source IP address and instead lets the 
> UDP/IP stack select one.
> 
> If the interfaces don't overlap in any way, I don't see how it would be 
> possible for Asterisk to send messages with the wrong source IP address, 
> since it does not specify the source IP address at all. If this is occurring, 
> it must involve the operating system's IP stack in some fashion.

Yes, I still use quite a lot of IPtables tricks to overcome this issue.

/O
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Re: [asterisk-users] Forcing SIP trunk matching order?

[Olle E. Johansson]

11 jul 2012 kl. 00:26 skrev James Lamanna:

> On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson  wrote:
> 
>> No.
>> 
>> This is probably because you are using phone numbers as names of devices 
>> with type=friend in sip.conf.
>> That's generally a bad idea.
>> 
>> The SIP channel matches an incoming call this way:
>> 
>> 1. Take the From: user name and match with the list of type=user and 
>> type=friend
>> 2. Take the sender's IP and port and match with the list of peers
>> 3. Send the call to the context defined in the [general] section of sip conf
>> 
>> In Asterisk 1.4 and hopefully 1.8 the last peer in sip.conf will match 
>> first. In 1.8 the internal strcutures
>> was changed, but I hope that this functionality was retained. We had a 
>> dicussion about it, but I personally
>> haven't tested the result. One needs to know the matching order, so if 1.8 
>> doesn't behave that way, we need
>> to fix it.
>> 
>> The recommended way is to NOT use anything that likely will end up as a 
>> caller ID as names
>> of devices in sip.conf. The name is whatever you have within square brackets 
>> above definitions
>> of type=friend or type=user. The username= option is another option, not the 
>> name of the device.
>> 
>> The quick way to solve your problems is to stop using type=friend and start 
>> using type=peer
>> instead.
> 
> Hi Ollie,
> 
> You are correct, I do have callerID-type names as accounts in sip.conf.
> The hosts are set to dynamic. Is this a problem with type=peer?
As stated above, peers only match on IP+port for incoming calls.

> 
> Would the deny/allow suggestion posted earlier also work with type=friend?

Deny/allow is a different thing and doesn't really affect matching. It is 
applied AFTER matching, not during or before.

Cheers,
/O
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Re: [asterisk-users] seems like call is picked and returned to me

[Olle E. Johansson]

9 jul 2012 kl. 15:24 skrev Sergio Serrano:

> Hi all
> 
> I hope that someone of you can solve this. Right now I'm stuck!
> I'm using asterisk with some SIP extensions. Basically I want to
> establish a call between desktop voip phone (ext 181) and embedded sip
> system (ext 182)
> 
> All I can see in CLI is:
> == Using SIP RTP CoS mark 5
>-- Executing [182@default:1] Dial("SIP/181-000a", "SIP/182")
> in new stack
>  == Using SIP RTP CoS mark 5
>-- Called SIP/182
>-- SIP/182-000b is ringing
>-- SIP/182-000b is making progress passing it to SIP/181-000a
>-- SIP/182-000b answered SIP/181-000a
>-- Remotely bridging SIP/181-000a and SIP/182-000b
>  == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a'
> 
> Seems like extension 182 (called ext) is getting call and passing them
> another time to me 181 (origin call)
> I've try it with siemens pbx and works as expected
> 

It's very hard to see what's happening without seeing the SIP logs. You just 
see that something went wrong in the process of setting up the bridge.

/O
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Re: [asterisk-users] Rookie / sip and extensions

[Olle E. Johansson]

7 jul 2012 kl. 21:07 skrev Mikhail Lischuk:

> Thomas Perron писал 07.07.2012 21:48:
> 
>> exten => s,n,Dial(SIP/16175551212)
>> 
>> 
>> sip.conf
>> [general]
>> ;register => 125010155:funnyti...@sip3.voipvoip.com/125010155
>> register => 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
>> ;
>> [incoming]
>> username=125010155
>> 
> I dont know what you are trying to do, but:
> 
> 1) Peer doesn't have to be the same name as context. Change [incoming] in 
> sip.conf to something like [voipvip] - it will be easier later when you have 
> more peers.
> 
> 2) What is 16175551212 ? You don't have such peer in sip.conf. If it's a 
> number, Dial should be SIP/peer/number, for example SIP/voipvip/617 or 
> whatever you want to dial
> 
> 3) If you've posted your real password here - I strongly suggest you change 
> it right now


Please note that the account name is the name within square brackets. 
The username= option (now renamed to defaultuser= ) is a very different thing, 
and NOT the username of the account.

/O
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Re: [asterisk-users] sip.conf and binaddr issue

[Olle E. Johansson]

6 jul 2012 kl. 23:18 skrev Felix Salfelder:

> Hi there.
> 
> i am seriously stuck with an asterisk and sip problem.
> 
> the following sip.conf works with respect to some_peer:
> 
> [general]
> bindaddr = x.y.z.w
> nat = no
> 
> [some_peer]
> type=peer
> host=somehost
> secret=somesecret
> some other
> unrelated options
> 
> here x.y.z.w is the ip address of the interface pointing to the network
> containing somehost. more precisely its the address of tun0 and route -n
> prints
> Destination Gateway Genmask Flags Metric RefUse Iface
> [..]
> x.y.z.0 0.0.0.0 255.255.255.0   U 0  0  0   tun0
> [..]
> 
> here 'it works' implies that i have to change and reload sip.conf after
> ifup tun0, or anything that forces tun0 to go down, like my dsl
> provider. also, the bindaddr line is suboptimal for the other peers...
> 
> the same thing -- without the bindaddr part -- doesnt work. more
> precisely it almost works. its just incoming sound that doesnt. this
> must have something to do with how asterisk picks up interface addresses
> and communicates them to the peer in question. inspecting the packages
> sent to somehost, gave me the impression that asterisk uses the ip
> adress of ppp0 (a dsl modem) instead.
> 
> how am i supposed to tell asterisk to use tun0 as the interface for
> [some_peer] so i can remove the bindaddr line? i've found many
> nat-related options in the manual, but there is no nat involved here.
> also, i couldnt find anything similar to "iface=tun0", although the sip
> dialogue apparently relies on ip adresses and routing.
> 
> this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course
> i'm going to switch to whatever you might suggest.
> 
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces with different IP networks,
Asterisk will send from the wrong IP on some of the interfaces.

Sorry,
/O


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Re: [asterisk-users] Timer1 RFC and SIP.CONF

[Olle E. Johansson]

6 jul 2012 kl. 09:29 skrev Elliot Murdock:

> Hello,
> 
> Thank you for the clarification.
> 
> Just a few questions:
> 1. What is the Timer1 used for?
Timer1 is the base for many other SIP timers and it's an estimate of the 
roundtrip time for a packet
between two SIP devices or servers. TimerB is based on T1, like the retransmit 
timers.

> 
> 2. Since timerb is for all responses, final and provisional, the
> comment in sip.conf perhaps should point that out instead of implying
> only for provisional responses: "If a provisional response is not
> received in this amount of time, the call will autocongest"
Yes, that should propably change.

/O
> 
> Thanks,
> Elliot
> 
> On Thu, Jul 5, 2012 at 12:31 PM, Olle E. Johansson  wrote:
>> 
>> 4 jul 2012 kl. 13:32 skrev Elliot Murdock:
>> 
>>> Hello,
>>> 
>>> I am trying to get clarity with the sip.conf timer configuration.  The
>>> current configuration states:
>>> 
>>> ;--- SIP timers
>>> 
>>> ; These timers are used primarily in INVITE transactions.
>>> ; The default for Timer T1 is 500 ms or the measured run-trip time between
>>> ; Asterisk and the device if you have qualify=yes for the device.
>>> ;
>>> ;t1min=100  ; Minimum roundtrip time for messages
>>> to monitored hosts
>>>   ; Defaults to 100 ms
>>> ;timert1=500; Default T1 timer
>>>   ; Defaults to 500 ms or the measured 
>>> round-trip
>>>   ; time to a peer (qualify=yes).
>>> ;timerb=32000   ; Call setup timer. If a provisional
>>> response is not received
>>>   ; in this amount of time, the call
>>> will autocongest
>>>   ; Defaults to 64*timert1
>>> 
>>> However, according to RFC 3261:
>>> 
>>> (EXCERPT 17.1.1.1)
>>> T1 is an estimate of the round-trip time (RTT), and
>>>  it defaults to 500 ms.  Nearly all of the transaction timers
>>>  described here scale with T1, and changing T1 adjusts their values.
>>>  The request is not retransmitted over reliable transports.  After
>>>  receiving a 1xx response, any retransmissions cease altogether, and
>>>  the client waits for further responses.  The server transaction can
>>>  send additional 1xx responses, which are not transmitted reliably by
>>>  the server transaction.  Eventually, the server transaction decides
>>>  to send a final response.
>>> 
>>> (EXCERPT 13.2.2.4 2xx Responses)
>>> The UAC core considers the INVITE transaction completed 64*T1 seconds
>>>  after the reception of the first 2xx response.
>>> 
>>> According to the RFC, the 64*t1 timeout is not for provisional
>>> responses, but for final responses.  This seems to be in contradiction
>>> to what is stated in the sip.conf file.
>> 
>> Unless you have PRACK support, which Asterisk not yet has, there's
>> no timeout in the SIP protocol for provisional responses more than
>> the need to update your provisional response at least every minute
>> not to hit a 3 minute timeout in the SIP transaction state in a proxy.
>> 
>> Now, the timerb is used to get ANY response from a server out there,
>> including provisional responses. If we don't get ANY response within
>> TIMERB, the SIP transaction dies and in a UA with a transaction
>> layer we generate a local 408 timeout. In Asterisk, we congest the call.
>> 
>> So the 64*T1 is for any response, including final response. It's there
>> to decide whether or not you have intelligent SIP life forms handling
>> your SIP request in the network universe.
>> 
>> I hope this clears up your confusion.
>> 
>> Regards,
>> /Olle
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Re: [asterisk-users] Timer1 RFC and SIP.CONF

[Olle E. Johansson]

4 jul 2012 kl. 13:32 skrev Elliot Murdock:

> Hello,
> 
> I am trying to get clarity with the sip.conf timer configuration.  The
> current configuration states:
> 
> ;--- SIP timers
> 
> ; These timers are used primarily in INVITE transactions.
> ; The default for Timer T1 is 500 ms or the measured run-trip time between
> ; Asterisk and the device if you have qualify=yes for the device.
> ;
> ;t1min=100  ; Minimum roundtrip time for messages
> to monitored hosts
>; Defaults to 100 ms
> ;timert1=500; Default T1 timer
>; Defaults to 500 ms or the measured round-trip
>; time to a peer (qualify=yes).
> ;timerb=32000   ; Call setup timer. If a provisional
> response is not received
>; in this amount of time, the call
> will autocongest
>; Defaults to 64*timert1
> 
> However, according to RFC 3261:
> 
> (EXCERPT 17.1.1.1)
> T1 is an estimate of the round-trip time (RTT), and
>   it defaults to 500 ms.  Nearly all of the transaction timers
>   described here scale with T1, and changing T1 adjusts their values.
>   The request is not retransmitted over reliable transports.  After
>   receiving a 1xx response, any retransmissions cease altogether, and
>   the client waits for further responses.  The server transaction can
>   send additional 1xx responses, which are not transmitted reliably by
>   the server transaction.  Eventually, the server transaction decides
>   to send a final response.
> 
> (EXCERPT 13.2.2.4 2xx Responses)
> The UAC core considers the INVITE transaction completed 64*T1 seconds
>   after the reception of the first 2xx response.
> 
> According to the RFC, the 64*t1 timeout is not for provisional
> responses, but for final responses.  This seems to be in contradiction
> to what is stated in the sip.conf file.

Unless you have PRACK support, which Asterisk not yet has, there's
no timeout in the SIP protocol for provisional responses more than
the need to update your provisional response at least every minute
not to hit a 3 minute timeout in the SIP transaction state in a proxy.

Now, the timerb is used to get ANY response from a server out there,
including provisional responses. If we don't get ANY response within
TIMERB, the SIP transaction dies and in a UA with a transaction
layer we generate a local 408 timeout. In Asterisk, we congest the call.

So the 64*T1 is for any response, including final response. It's there
to decide whether or not you have intelligent SIP life forms handling
your SIP request in the network universe.

I hope this clears up your confusion.

Regards,
/Olle
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Re: [asterisk-users] port 5060 is blocked by ISP

[Olle E. Johansson]

1 jul 2012 kl. 09:48 skrev Leandro Dardini:

> Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet 
> is using TCP.

That's not correct. SIP supports multiple transports, including TCP. Not all 
implementations support TCP though.

/O
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Re: [asterisk-users] Forcing SIP trunk matching order?

[Olle E. Johansson]

28 jun 2012 kl. 23:25 skrev Duncan Turnbull:

> Hi James
> 
> On 29/06/2012, at 6:19 AM, James Lamanna wrote:
> 
>> Hi,
>> I have a bunch of different customers on an Asterisk Box (the PBX).
>> This Asterisk Box is behind another Asterisk box that provides a PSTN
>> connection.
>> Up to this point I've been using IAX between the 2 Asterisk boxes, but
>> I would like to use SIP instead.
>> After doing some testing I have the following issue.
>> 
>> If customer A calls customer B, but the call goes out through the PSTN
>> and comes back in, the call is rejected at the PBX because it wants
>> authentication.
> 
> This raises the question of how calls come in from the PSTN in the first 
> place. I am guessing you route them out in order to bill them? I am guessing 
> you have more than one SIP trunk between the two boxes for different purposes 
> and what you are saying is the authentication is falling to the lowest common 
> denominator.
> 
> In this situation you could separately register two lines, raise your level 
> of security ( insecure = no option), you need to make sure there is an 
> incoming context that matches the supplied user name otherwise it will fall 
> back to the ip address and whatever context that matches
> 
> 
>> I can guess that this must be because it matches the "To" address to
>> the friend SIP trunk that provides registration for the phone.
>> (All phones are setup as type=friend, host=dynamic). Is there any way
>> to force matching against the peer SIP trunk to PSTN so as to not
>> require authentication for calls
>> coming in from the PSTN server?
>> 
> Using permit and deny directives for the ip address should help here

No.

This is probably because you are using phone numbers as names of devices with 
type=friend in sip.conf. 
That's generally a bad idea.

The SIP channel matches an incoming call this way:

1. Take the From: user name and match with the list of type=user and type=friend
2. Take the sender's IP and port and match with the list of peers
3. Send the call to the context defined in the [general] section of sip conf

In Asterisk 1.4 and hopefully 1.8 the last peer in sip.conf will match first. 
In 1.8 the internal strcutures
was changed, but I hope that this functionality was retained. We had a 
dicussion about it, but I personally
haven't tested the result. One needs to know the matching order, so if 1.8 
doesn't behave that way, we need
to fix it.

The recommended way is to NOT use anything that likely will end up as a caller 
ID as names 
of devices in sip.conf. The name is whatever you have within square brackets 
above definitions
of type=friend or type=user. The username= option is another option, not the 
name of the device.

The quick way to solve your problems is to stop using type=friend and start 
using type=peer
instead.

Regards,
/O
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Re: [asterisk-users] SIP over SSL TCP or SRTP?

[Olle E. Johansson]

22 jun 2012 kl. 21:59 skrev Bruce B:

> Thanks. Want to secure everything and anything possible. 
> 
> 1- Can both  SIP over TLS  and SRTP work in conjunction to each other?
Yes. As Kevin said, SIP over TLS only secures the signalling. And it secures it 
hop-by-hop so every server in the middle
can access the content. The signalling should be hidden from other Wifi users, 
even if it's not hidden all the way between
caller and callee. In the signalling you specify how to exchange the actual 
media. To have secure signalling with TLS
doesn't necessarily mean that them media (audio/video/text) is secured. The 
media is secured with Secure RTP or SRTP,
which means that every audio packet is encrypted.

> 2- Is SIP over TLS a package or added on module that can be installed from 
> Digium Asterisk repository?
It's part of the current Asterisk SIP stack, but still marked as experimental 
as it has a number of known issues that needs to be fixed
in order to put this in production use in larger sites and networks. You will 
have to test it to make sure it works for you.

"Experimental" status means that the configuration options may change in a 
coming release without being backwards
compatible. The TLS part has been experimental in many releases without anyone 
putting any funding towards
fixing it. I guess serious use of TLS is done not with Asterisk but with a SIP 
proxy like Kamailio or OpenSIPS in
front of Asterisk.

> 3- SRTP takes care of the RTP and makes it secure so that MITM type sniffing 
> is not possible?
Yes, provided that the media encryption key exchange is secured. Today, the key 
exchange is done in SIP messaging,
which is why you also want SIP over TLS.

Regards,
/Olle
> 
> Regards,
> 
> 
> 
> On Fri, Jun 22, 2012 at 2:39 PM, Kevin P. Fleming  
> wrote:
> On 06/22/2012 12:56 PM, Bruce B wrote:
> 
> Which one of these ensures that SIP packets are sent and received in a
> secure format so that users using public wifi don't allow MITM type of
> attacks or others can't read the plaintext SIP packet info. VPN is not
> an option. Looking for 2nd most secure to VPN.
> 
> SIP over TLS (what used to be called SSL) is what secures the SIP signaling. 
> SRTP is for securing media streams.
> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
> 
> 
> 
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[asterisk-users] Community event: Open Source Realtime Dinner in Barcelona - June 13th

[Olle E. Johansson]

Hello!

I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in 
June. During this week, I will organize a dinner for everyone working with or 
interested in Asterisk, Kamailio and other Open Source platforms for realtime 
communication. It's June 13th somewhere in Barcelona - location will be 
announced later. You pay our own dinner (unless we can find sponsors) and enjoy 
the geeky company for free!

To join the event, use this Facebook event 
https://www.facebook.com/events/307548349321608/

See you in Barcelona!

/O

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Re: [asterisk-users] Invite + decreasing sequence number => 500 Error?

[Olle E. Johansson]

16 apr 2012 kl. 15:31 skrev Matthew Jordan:

> It's not a bug - decrementing the CSeq header field value is directly in
> violation of RFC 3261.  From section 22.2:
> 
>   When a UAC resubmits a request with its credentials after receiving a
>   401 (Unauthorized) or 407 (Proxy Authentication Required) response,
>   it MUST increment the CSeq header field value as it would normally
>   when sending an updated request.

This only applies to the same dialog. The question here is if it is the 
same dialog. If it is, then the server indeed has a bug.

Check the Call-ID and the from tag of both requests.

/Olle
> - Original Message -
>> From: "Benoit Panizzon" 
>> To: asterisk-users@lists.digium.com
>> Sent: Monday, April 16, 2012 7:12:09 AM
>> Subject: [asterisk-users] Invite + decreasing sequence number => 500 Error?
>> 
>> Hi out there
>> 
>> We have a strange Problem here with invites.
>> 
>> We observe this SIP conversation.
>> 
>> C3 PBX <-> Asterisk
>> 
>> Case 1. Sequence Numer always increasing:
>> 
>> => Invite
>> <= 401 Unauthenticated
>> => Invite+auth with sequence number > previous Invite.
>> <= 100 Trying etc. Works OK.
>> 
>> Case 2. Sequence Number decreasing.
>> 
>> => Invite
>> <= 401 Unauthenticated
>> => Invite+auth with sequence number < previous Invite.
>> <= 500 ERROR
>> 
>> I was browsing the SIP rfc and I cannot find any clue if in this case
>> the
>> sequence numbers must be increasing (the C3 PBX is wrong) or if I
>> have sumbled
>> over an asterisk bug.
>> 
>> Is there anyone who knows?
>> 
>> Benoit Panizzon
>> --
>> I m p r o W a r e   A G-
>> __
>> 
>> Zurlindenstrasse 29 Tel  +41 61 826 93 07
>> CH-4133 PrattelnFax  +41 61 826 93 02
>> Schweiz Web  http://www.imp.ch
>> __
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Re: [asterisk-users] Asterisk as register server through OpenSIPS

[Olle E. Johansson]

9 jan 2012 kl. 09:02 skrev Ronald Cepres:

> Hi all,
> 
> I've been trying to register a SIP user agent to an Asterisk server using 
> OpenSIPS as SIP router. The functionality is working fine. However, Asterisk 
> uses the IP address of the OpenSIPS server as the peer IP address. How can I 
> use the original IP address of the peer without changing the peer's nat=yes?
> 
> 
> I appreciate any kind of help. Thanks!

You propably have NAT=yes in Asterisk. 

If you turn that off, Asterisk will save the contact provided by the phone 
which will point directly to the phone, bypassing the OpenSIPS proxy. In order 
to get Asterisk to use the OpenSIPS proxy outbound as well you need to define 
it as an outbound proxy. 

Now, you have to configure NAT support in OpenSIPS since it's the first hop 
seen from the phone.

/O


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Re: [asterisk-users] VUC: AstLinux 1.0.0 release

[Olle E. Johansson]

17 dec 2011 kl. 10:11 skrev Darrick Hartman:

> The AstLinux Team is happy to announce the release of AstLinux 1.0.0.  This 
> release includes significant changes and improvements over past releases.  
> Specific upgrade and new installation instructions are available at:  
> http://www.astlinux.org 
> 
> Some of the highlights include:
> 
> * Using eglibc instead of uClibc. This allows binary compatibility with 
> add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc).
> * Newer Kernel which better supports newer hardware
> * Support for Jabber/Gtalk
> * Removed mISDN support (the zaphfc DAHDI driver is included for single port 
> ISDN cards)
> 
> A full changelog is available on the release pages.  We provide versions with 
> Asterisk 1.8 and 1.4.  
> 
> Because this is a major version change, there are some special considerations 
> when upgrading.  Please read the instructions very carefully to ensure no 
> step is skipped.
> 
> http://doc.astlinux.org/userdoc:upgrade-0.7 
> 
I happily noticed support of IPv6! Congratulations to the new release.

/O

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Re: [asterisk-users] Which device auto-registered an extension?

[Olle E. Johansson]

16 dec 2011 kl. 18:12 skrev Barry Miller:

> On Fri, Dec 16, 2011 at 05:02:11PM +0100, Olle E. Johansson wrote:
>> 
>> 16 dec 2011 kl. 02:03 skrev Barry Miller:
>> 
>>> So is there a way for the dialplan to determine which device caused SIP to
>>> auto-register an extension?
>> 
>> Not really, unless someone else can come up with something. 
>> 
>> In Asteirsk, the extension hints are the connection from the dialplan to a 
>> device,
>> used for subscriptions and blinking lamps.
>> 
>> exten => 543,hint,SIP/devabc
>> 
>> then you can use
>> 
>> exten=> _5XX,DIAL(/${HINT})
>> 
>> Which opens up the question on how you enter all the hints...
>> 
>> I know Tilghman added something clever recently to new versions of Asterisk, 
>> but I 
>> haven't used it myself so I can't describe how it works.
> 
> Thanks.  I like to keep devices and extensions separate, for security 
> reasons, and it's a
> pain having to define the association in different places, especially when 
> sip.conf seems
> like the logical place to do this.  Do you think if I were to code up a 
> function like
> AUTOREG([context,]exten) that returned 1 for auto-registered extensions, 0 
> otherwise, and
> set a channel variable to the registering device name, it might be useful 
> enough to be
> accepted as a feature?


I would assume that to extend regexten with reghint and have a similar 
functionality
where we add a hint to an extension at registration would work nicely within 
the 
current architecture.

/O
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Re: [asterisk-users] Best PBX for Call Centers?

[Olle E. Johansson]

15 dec 2011 kl. 19:33 skrev Tarek Sawah:

> 
> Hello List,
> I have customer with a 40 Agents call center. and is looking to install a PBX 
> switch that can serve those agents.
> As per my experience i suggested Asterisk as i have tested it with Call 
> Centers, however he has been advised not to use it although his provider is 
> using Asterisk to send him calls. He has been advised to use Sippy which they 
> claim is more stable than Asterisk. 
> i'm not an expert with Sippy so i'm looking for a piece of an advise here.. 
> if i'm doing an Asterisk Vs Sippy comparison. can anyone help?
> Regards
> 

We have installed Asterisk 1.4 in Call centers ranging up to 400 agents on one 
PBX. It's clearly stable and doesn't crash.

I would recommend talking with this service provider again.


/O

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Re: [asterisk-users] Which device auto-registered an extension?

[Olle E. Johansson]

16 dec 2011 kl. 02:03 skrev Barry Miller:

> Hi all,
> 
> In sip.conf:
>  [general]
>  regcontext = autoreg
> 
>  [devabc]
>  regexten = 543
> 
> creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc
> registers.  But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the
> dialplan, because there's no device SIP/543.  Now I know I can add a line
> like "exten=> 543,2,Dial(SIP/devabc)" for each and every device that uses
> regexten, but it would be a lot cleaner to be able to use something like
> Dial(SIP/${WHAT_GOES_HERE?}) instead.
> 
> So is there a way for the dialplan to determine which device caused SIP to
> auto-register an extension?

Not really, unless someone else can come up with something. 

In Asteirsk, the extension hints are the connection from the dialplan to a 
device,
used for subscriptions and blinking lamps.

exten => 543,hint,SIP/devabc

then you can use

exten=> _5XX,DIAL(/${HINT})

Which opens up the question on how you enter all the hints...

I know Tilghman added something clever recently to new versions of Asterisk, 
but I 
haven't used it myself so I can't describe how it works.

/O

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Re: [asterisk-users] SIP Trunk

[Olle E. Johansson]

16 dec 2011 kl. 11:29 skrev James Courtier-Dutton:

> Hi,
> 
> I have a situation where unfortunately, I cannot use IAX for trunks,
> and need to instead use SIP trunks.
> Is there any way to fit the voice data from more than one simultaneous
> phone call into a single IP packet over the SIP trunk.
> I believe this is possible with IAX trunks, but I don't know how to do
> it for SIP trunks.
> 
Yes there is a proposal for RTP trunking, but it's not implemented in Asterisk.

/O


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[asterisk-users] SIPit 29 in Monaco - interoperability by hard work

[Olle E. Johansson]

Friends,

SIPit is an event organized by the SIP Forum and partners. It has been running 
for 15 years twice a year, making sure that SIP clients and servers 
interoperate. By testing, we also find issues with the myriad of RFCs in this 
area and correct them. Testing interoperability is important.

The first time I brought Asterisk to SIPit in Stockholm many years ago I was 
terrified. The SIP stack back then was, well, peculiar. It worked with some SIP 
phones for basic calls, but not much more. During the tests I learned a lot, 
got a lot of help from friendly engineers and fixed a large amount of bugs. 
Afterwards I had a list of todo's that kept me busy for quite some time. It 
helped Asterisk leaping forward in the SIP area. We've grown since then and 
Digium, as a member of the SIP Forum, hosted a SIPit in Huntsville a while ago, 
testing both Asterisk and the new baby, Asterisk SCF. I have hosted a SIPit 
here in Sweden. The Asterisk eco system  believes in interoperability and we 
work hard to stay interoperable with the world of SIP.

This year at SIPit we'll run all kinds of tests. I will personally focus on 
security and IPv6 tests. Together with the Kamailio/SIP-router team, I've built 
automatic tests in these areas. Hopefully we can complete them and make them 
public. (We have a few cool Allison prompts for them too!)  I guess I will find 
a few new bugs as well :-)

So why am I writing to the Asterisk-users mailing list about this? It's not 
about showing off, it's about explaining how interoperability happens. 
Customers need to require interoperability and open standards, not accept any 
vendor lock-ins. Vendors and Open Source developers need to take the lead and 
work together to get interoperability. Customers need to test. 

So if you are a developer of SIP products that interact with Asterisk - please 
register for SIPit today. If you are a customer, please ask your vendors if 
they participate in SIPit. If you have a lot of knowledge about Asterisk, 
please register and come test Asterisk 10. 

If you are a happy Asterisk user, please understand that the great level of 
interoperability that we have in Asterisk, communicating with all kinds of 
devices and servers out there, just did not happen by accident. It required a 
lot of work by a large group of developers. And you benefit.

I'll try to keep you posted about the progress of the tests. I can't name 
specific vendors that participate, it's part of the deal. But we can talk about 
the test results, the state of certain features. Personally I hope that we can 
succeed in many more tests with TLS and SRTP this event. And we'll have some 
fun with IPv6 and dual stack implementations as well.

Thanks for reading!

Have a nice weekend!

/O
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Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

20 sep 2011 kl. 15:34 skrev Danny Nicholas:

> 
> Just my .02 - fix Originate since the "Original Asterisk" book, page 125
> paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to

In the first edition, page 82, it actually says ""When a call enter a context 
without a specific destination extension, they are handled automatically by the 
s extension". Which is correct. It continues "(The s stands for start, as most 
calls start in the s extension)" which is very wrong.

In the edition you have, page 125, the "most calls" part is deleted and the 
text explains that "this is where a call will start if no extension information 
was passed with the call". 

So they got it right in the end :-)

/O
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Re: [asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

20 sep 2011 kl. 15:34 skrev Danny Nicholas:

> Just my .02 - fix Originate since the "Original Asterisk" book, page 125
> paragraph 1 says "s" = "start".  If "s" is not really "start", I'm going to
> scrap my 3+ years of dialplan writing and change all of my simple dialplans
> to read exten=> start,1,blah instead of exten => s,1,blah.  To me exten=>
> s,1,blah is more intuitive and less vulnerable than exten => _X.,1,blah.

I am sorry that the "Original Asterisk" book was wrong and do hope that they 
corrected that part in later editions.

I don't think any official docs have pointed out that "s" was anything else 
than a default extension for situations where there is no extension given.

Using "start" makes your dialplans much easier to read :-) and makes them more 
secure as no app will end up there by accident, which may happen in your 
current systems.

/O
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Re: [asterisk-users] DTMF problem

[Olle E. Johansson]

19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:

> This DTMF problem has always been there and there is no real solution for it, 
> other than using those expensive PBX systems like that from Avaya, Cisco, 
> etc. This problem happens when you are sending inband DTMF tones. Via 
> softphone you are sending out-of-band DTMF which is basically SIP messages.

Just to correct the latest part of your statement:

The default way to send DTMF in SIP calls is using DTMF as a codec called 
telephony-event in the RTP stream. This sends DTMF as events. Most hard and 
soft phones support this - usually called RFC2833 DTMF mode. Asterisk supports 
it by default. 

Sending DTMF in the audio usually gets messy when using an IP network. 
Especially if you use codecs that compress the audio. I do recommend you to use 
RFC2833. We have built very large IVR services and have no issues with DTMF 
being received in Asterisk so it's doable.

There are other issues with Asterisk DTMF, but that's another issue :-)

/O




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Re: [asterisk-users] single registration per user

[Olle E. Johansson]

18 sep 2011 kl. 22:23 skrev Catalin S.:

> Hello Eric,
>  
> Is about outgoing calls from multiple devices with the same username at aprox 
> same time. The overwritten is for incomming calls. I want to prevent using 
> the same account in multiple devices at same time. The solution with IP will 
> not apply because users may be behind nat or will change everytime multiple 
> access points. Do you have any other clues?

There is no real good way to prevent this. How can Asterisk now which 
registration that is the valid one? If a device reboots and gets a new IP from 
DHCP, we do not want to prevent that new registration to prevent the old one 
from another IP, but the very same device. There's no device ID used in the 
registration, only the SIP account. 

This also applies to OpenSER/kamailio/OpenSIPS. We can prevent multiple 
simultaneous registrations in those, but that will also mean that phones that 
reboot will be blocked until all registrations expire in the server.

/O


Edvina SIP Masterclass covering SIP, Asterisk & Kamailio - Oxford, UK, Nov 
7-11. *  http://www.telespeak.co.uk
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[asterisk-users] Fixing an old bug related to extension "s" - feedback wanted

[Olle E. Johansson]

Friends,

While working with the manager interface, I noticed that an originate action to 
a non-existing extension had a strange behaviour. Instead of generating an 
error, which would happen in most VoIP channels and Dahdi, Asterisk started 
looking for extension "s" as a "fallback". 

For as long as I've worked with Asterisk, the definition of extension "s" has 
been that it is used when *NO EXTENSION* has been given (and in the macro 
command). There are two good examples - immediate answer in Dahdi and calling a 
SIP domain without a username part - like "sip:digium.com". In my trainings I 
always repeat (with a loud voice) that extension "s" is *NOT* a wildcard.

Obviously this behaviour is a bug. It's been around for a long time and has 
been hidden by most apps and channel drivers that handle a bad extension in a 
correct way and report errors before the PBX is started in order to handle the 
channel.

The question is - how do we fix this? There might be applications out there 
that depend on this buggy behaviour.

What I've proposed are two separate fixes:

1) Change the manager Originate action


In Asterisk 1.8, there will be a warning if an extension given doesn't exist, 
but the behaviour will not change. A flag in Asterisk.conf [compat] section 
will be implemented so that you can change this behaviour and get an error 
response in manager if the extension does not exist.
In Asterisk 10 the error response will be the default behaviour. If an 
application using AMI needs a fallback, it needs to be controlled by the 
application. It needs to know that an extension does not exist and that the 
call can't be fulfilled.

2) Change the PBX core
===

The bug actually exists in the PBX core, in ast_pbx_start(). We will not change 
this in Asterisk 1.8. 

In Asterisk 10, the core pbx will report that the extension does not exist and 
no longer fall back to s in current context or s@default. This will, as we see 
it now, not affect most channel drivers and thus most dialplans. If you rely 
heavily on the originate function (AMI, CLI and dialplan)  and use the fallback 
behaviour, you will need to modify your dialplans.

Final question
===

My question now is what you think about these changes. Do you need a flag for 
Asterisk 10 to revert to the old behaviour? Is this bug something you actually 
rely on in your application?

Thanks for your response!

/O


Edvina SIP Masterclass covering SIP, Asterisk & Kamailio - Oxford, UK, Nov 
7-11. *  http://www.telespeak.co.uk



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Re: [asterisk-users] Overlap SIP dialing

[Olle E. Johansson]

8 sep 2011 kl. 17:26 skrev Andrew Latham:

> On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson  wrote:
>> 
>> 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:
>> 
>>> Honestly, I'm not really sure that there is a practical solution here. ISDN 
>>> overlap dialing was intended for 'dumb' phones, and SIP phones aren't 
>>> 'dumb' :-)
>> 
>> That's a quote that goes to my quote storage layer.
>> 
>> /O ;-)
>> --
> 
> I want a t-shirt   "SIP phones aren't 'dumb' :-)"
> Overlap dialing has very limited use, however I found it helpful when
> testing integration with other PBX/VM/PSTN connections.

Yes, but the solution is not 484, but as Kevin stated to answer the darn call 
on the SIP side and provide a dial tone from the other side. And yes, we do 
have dial tones on ISDN PRI trunks...

/O
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Re: [asterisk-users] Overlap SIP dialing

[Olle E. Johansson]

8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:

> Honestly, I'm not really sure that there is a practical solution here. ISDN 
> overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' 
> :-)

That's a quote that goes to my quote storage layer.

/O ;-)
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Re: [asterisk-users] Overlap SIP dialing

[Olle E. Johansson]

7 sep 2011 kl. 16:20 skrev Andrew Latham:

> 
> 
> On Wednesday, September 7, 2011, Olle E. Johansson wrote:
> 
> 7 sep 2011 kl. 15:59 skrev Daniel Tryba:
> 
> > Looking at the history of the list I don't expect any answer but lets
> > try anyway:
> >
> > Does anybody use overlap dialing from SIP devices to asterisk? Does
> > anybody have a working example?
> 
> To add to your question: Does anyone have a phone that supports this properly?
> 
> /O
> 
> Yup, I have a few...  http://wiki.snom.com/Settings/overlap_dialing 
> 

Great. Haven't seen this - thank you.

The whole concept is interesting. Suppose the call forks and one UA answers 
with 484, another with 486 and another with 180 ringing. What are you supposed 
to do? I think there's a problem with the RFC 3261 here and don't know if it's 
been clarified.

Now - in the case of Asterisk if we call out to two devices from the dialplan 
and one responds with 484 and another with 180 ringing - what happens in 
Asterisk?

/O
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Re: [asterisk-users] Overlap SIP dialing

[Olle E. Johansson]

7 sep 2011 kl. 15:59 skrev Daniel Tryba:

> Looking at the history of the list I don't expect any answer but lets
> try anyway:
> 
> Does anybody use overlap dialing from SIP devices to asterisk? Does
> anybody have a working example?

To add to your question: Does anyone have a phone that supports this properly?

/O
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Re: [asterisk-users] Set(CHANNEL(musicclass)=

[Olle E. Johansson]

6 sep 2011 kl. 22:30 skrev Leif Madsen:

> On 02/09/11 11:27 PM, Joseph wrote:
>> In asterisk 1.4 I had:
>> exten => s,n,Answer()
>> exten => s,n,SetMusicOnHold(default)
>> 
>> But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default)
>> (beside it is deprecated) as it is default.
>> In 1.6 and UP I think it is: Set(CHANNEL(musicclass)= Can somebody
>> explain what do they mean by "CHANNEL"?
> 
> CHANNEL() is a dialplan function. You're setting parameters for the current 
> channel by using that function. So instead of using a dialplan application 
> like you were before, you use the CHANNEL() function.
> 
> exten => s,1,NoOp()
> same => n,Set(CHANNEL(musicclass)=default)
> 
>> I could use just:
>> exten => s,n,MusicOnHold()
>> 
>> There is a lot of documentation on www.voip-info.org but sometimes it is
>> not clear which asterisk version it applies to :-/
> 
> (Another good reason to be reading the documentation on 
> https://wiki.asterisk.org/wiki instead :))
> 
...or update and help maintaining the voip-info documentation.

/O
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Re: [asterisk-users] Wanted a modified SIP message body

[Olle E. Johansson]

31 aug 2011 kl. 14:42 skrev Kevin P. Fleming:

> On 08/31/2011 02:46 AM, Jaime Lozano wrote:
>> Hello,
>> I agree with you, I'm not explaining the problem in a proper manner,
>> because of my lack of Asterisk knowings. I send the Wireshark captures.
>> 
>> 3com telephones take the timezone TZ:7200 from the 3Com PBX to show the
>> time right. But what if I want a 3Com telephone to work with Asterisk
>> PBX? Then the telephone time is wrong, 2 hours lower. It seems 3Com
>> telephones need the TZ:7200. 3Com telephones work with Asterisk and it
>> is great, but we would like to log the calls.
> 
> OK, so the first clarification is that you are talking about responses to 
> REGISTER requests specifically, not all responses to all requests. That's 
> good :-)
> 
> On to the meat of the issue... indeed, the '200 OK' response to a REGISTER 
> request does not normally have a message body; nothing in the SIP RFCs even 
> suggests that there would be one (although it's certainly allowed should the 
> registrar want to include it) or what would be present in it.
> 
> As has been previously replied here, there is no facility in Asterisk to 
> include a message body in a REGISTER request response, so providing one will 
> definitely require source code modifications. They wouldn't be terribly 
> difficult, but they would only be applicable to these particular phones, 
> which reduces the benefit of making the changes to the community at large.
> 
> With that said... it's certainly possible to do this, but it's going to take 
> some non-trivial code changes. The REGISTER handling code does not use any of 
> the methods that exist in chan_sip to add message body content to its 
> responses, it uses simpler methods that assume there won't be a message body.
> 
> In addition, this mechanism is really pretty broken anyway; the server would 
> have to know where each phone is physically located in order to be able to 
> provide the correct TZ value to it, and would have to be updated if the phone 
> is moved. Not an ideal situation.

The RFC states that a phone could use the Date: header in the response to set 
the local time in the device. It's always in GMT which makes it stupid to add a 
time zone any where. 

-1 for this implementation.

/O
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Re: [asterisk-users] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]

[Olle E. Johansson]

29 aug 2011 kl. 15:05 skrev Kevin P. Fleming:

> On 08/28/2011 01:56 AM, Tzafrir Cohen wrote:
>> On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote:
>>> Hi
>>> 
>>> I've just added direct support for AMI to a forthcoming version of
>>> TBDialOut, a Thunderbird extension for dialling direct from
>>> Thunderbird's address book. If anyone fancies testing it I'd be grateful
>>> for any feedback. If you feel like casting a critical eye over the code,
>>> or doing some translating, even better.
>>> 
>>> AMI support is available in TBDialOut 1.7.0pre1, which can be found
>>> either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development
>>> channel' at the bottom of the page at
>>> https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/
>> 
>> We already have a dialer script (sent to this list a while ago) so it's
>> good to see that this extension support that simpler option as well (I
>> don't use ThunderBird, as you can see. Some others in the office do use
>> it).
>> 
>> One followup question: I originate a call from a SIP phone to some
>> remote number. The problem is that the number will not show up properly
>> in the list of outgoing calls for the phone. Any idea how to fix this
>> (for whatever SIP phone)?
> 
> You aren't originating a call *from* the phone (that would require some sort 
> of API into the phone itself to make it place a call). You are originating a 
> call *to* the phone and also to another endpoint; as far as the SIP phone is 
> concerned, this is an incoming call.
> 
> I've never seen discussion of a desire to provide a method for an incoming 
> call to be treated as if the endpoint had placed the call itself in any of 
> the SIP discussion lists I frequent... so I'm pretty sure there's no standard 
> way to do this.
Oh, there is - REFER.

We could possibly implement sending a REFER request to the phone, something 
that is frequently used to do call setups from click-to-call apps. This is not 
something we do support in Asterisk today. I've implemented it using SIP 
libraries since Asterisk doesn't have to be involved in the REFER.

If you do ORIGINATE from the phone you have to be aware that Asterisk lacks 
some security framework here. An application that has ORIGINATE access can 
reach the whole dialplan. I have patches for that which needs to be moved 
forward. My proposal is to add a default context to manager accounts to put a 
limitation of destinations they can reach with originate and redirect AMI 
commands (which where the only ones I could come up with as dangerous in this 
regard).

/O
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Re: [asterisk-users] Wanted a modified SIP message body

[Olle E. Johansson]

26 aug 2011 kl. 14:06 skrev Jaime Lozano:

> Hello,
> In which file do I use SIPAddHeader()?
> Please consider that the packet goes from the PBX to the telephone, and what 
> I want is not a header because the "TZ: 7200\n" is in the *message body* not 
> in the *message header*.

That's no longer a SIP header, it's part of the SDP you want to change. You 
can't do that without changing the source code.

/O
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Re: [asterisk-users] Problem setting for incoming termination

[Olle E. Johansson]

12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:

> On 08/11/2011 02:03 AM, Jim Boykin wrote:
> 
>> We have difficulty setting up the incoming termination for our
>> clients. Both the ends are using asterisk.  The problem is unless we
>> use fromuser at client end, it does not work properly as expected.
>> 
>> Below is a configuration at our end. The problem is that whenever call
>> is received from the client, it goes to default context instead of
>> 'dallas' context. Also, the ${CDR(accountcode)} variable remains
>> empty. Now, If we set fromuser field at the client end, then
>> everything starts working, however, in that case, it overrides the
>> callerid.
> 
> This is a known and well-understood problem caused by the method that 
> Asterisk users for SIP authentication; the 'From' header in the incoming 
> INVITE is used *both* for determining which user is placing the call and for 
> Caller ID. As you note, if you have the real Caller ID in that header, then 
> Asterisk can't use it for matching to a user in sip.conf, and thus can't 
> authenticate the call properly.
> 
> The solution for this is to use 'sendrpid' on the sending end and 'trustrpid' 
> on the receiving end; this will configure Asterisk to transfer the Caller ID 
> information in a Remote-Party-ID (or P-Asserted-Identity, depending on the 
> version you are using) header, allowing the From header to be used solely for 
> authentication.

Or stop using type=user and type=friend, and stick to type=peer and ASterisk 
will only match on IP+port address.

/O
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[asterisk-users] Asterisk/Kamailio dinner in Madrid thursday next week - June 30th

[Olle E Johansson]

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - 
The SIP master class.

Maybe we can organize a voip nerd dinner on Thursday evening? If you're 
interested, please e-mail me off list and I'll send out more details later. 

Greetings
/Olle

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Re: [asterisk-users] asterisk fails when DNS or internet fails

[Olle E Johansson]

31 maj 2011 kl. 14.49 skrev Benny Amorsen:

> Jeff LaCoursiere  writes:
> 
>> Hasn't anyone managed to solve this with something better than a
>> caching DNS server, which seems to only last a short while?  What
>> exactly is going on that is failing?
> 
> If your recursive DNS server returns errors quickly rather than actually
> trying to look up the names, Asterisk works fine.
> 
> It is not a particularly nice workaround, but it does work... As long as
> Asterisk does not actually NEED the DNS information, but that can be
> most worked around with static configuration of IP addresses in sip.conf.
> 

Longterm we should really integrate an Asynchronus DNS library, like C-Ares.

I've been wanting to do that for years.

/O

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Re: [asterisk-users] SIP per-call heartbeat?

[Olle E. Johansson]

24 maj 2011 kl. 12.19 skrev Tony Mountifield:

> One of our customers has an Asterisk conference bridge connected to a
> SIP trunk from an ITSP. Yesterday, they had two inbound calls that
> didn't get hung up properly. From the tcpdump SIP trace that we have
> running continuously, I can see that no BYE was received by the bridge,
> and when some hours later the hangup was forced from the bridge end, the
> bridge sent a BYE to which it received a 481 Call Leg/Transaction Does
> Not Exist.
If the remote end send a BYE and doesn't receive a response, that bye will have 
to be retransmitted multiple times before it gives up. The SIP protocol 
includes retransmission over UDP, to cover up for packet loss. If it did not 
retransmit, you have other issues.

> 
> Since SIP is UDP, this situation must occur from time to time, and I
> wondered if it is possible to configure any kind of per-call SIP
> heartbeat so that a dead call could automatically be identified with a
> 481 response much sooner.

SIP session timers is what you need for that. Implemented in Asterisk 1.8.

/O
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Re: [asterisk-users] Sending call to specific IP address

[Olle E. Johansson]

23 maj 2011 kl. 23.36 skrev Paul Belanger:

> On 11-05-23 05:30 PM, Elliot Murdock wrote:
>> Hello,
>> 
>> I am wondering how to send a call to a specific IP address that is different
>> than the host of the URI.  For example, an invite to the URI is "
>> j...@phone.com" needs to be sent to the IP address 123.456.789.255, not to
>> the IP address of phone.com.
>> 
>> How is this done?
>> 
> Look at the 'Contact' header.
> 
I don't what Paul means here... YOu can surely define a peer and add an 
outbound proxy with the IP address... That way we won't overwrite the domain.  
I am not aware of a way of doing it in the dial string in the dial plan.

/O


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

5 maj 2011 kl. 18.30 skrev Ira:

> At 07:56 AM 5/5/2011, you wrote:
>> So how can we fix this?  How can we get more people involded?  What makes 
>> projects like FedoraTesting[3] and DebianTesting[4] popular?  How can the 
>> Asterisk project reproduce their success?
> 
> Well, it's not a lot of people willing to run beta software on their phone 
> system. Phones need to work and for most people they need to work perfectly 
> all the time. I'm one of those oddities that will always run beta software if 
> given the chance but my experience is that quite rare.
> 
>> As I've said before, I'm more then willing to help with answering questions 
>> about the testsuite or reviewing code that people want to get merged in.  We 
>> also have an IRC channel, #asterisk-testing available for people to join, 
>> ask question, idle, lurk, etc, or if you want to reply to this thread, feel 
>> free.  But get involved! :)
> 
> So I'm the person who has never been able to keep 1.8 alive on my system for 
> more than a minute or two and I've probably tried more than 10 different 
> betas and release versions. I posted a bug report which was closed in 
> minutes, I posted the problem on this list every few tries and zero response. 
> I tried to figure out mIRC. It's installed on my machine but I've never got 
> past that. I just don't get the instructions.
> 
> I know that all the people involved in the project are Linux heads, but some 
> of us, like me, have a Linux box only because of Asterisk and if you want my 
> help, you need to make being involved accessible and stop assuming we all 
> know what you know. I see the words, "jut post a bug report on Mantis" posted 
> all the time and I'm sure it means as little to others as it means to me. 
> Maybe there needs to be a web page somewhere, "Asterisk beta testing for 
> dummies" so that you can point us to so you don't have to answer the stupid 
> questions over and over.
> 
> I've beta tested enough and had enough beta testers to understand the kinds 
> of things that make it possible to get bugs fixed, but it's usually a very 
> small percentage of users that understand that.

Thanks for the feedback, Ira. It makes me very sad to hear what you say and I 
hope that we can get more resources from the community to assist in the process 
to make it more friendly. We want to get those bug reports. The one thing I 
hate to hear when I'm travelling at conferences is that "oh, I known that bug 
for a long time but did not bother to report it." 

Apologies for your experience with the bug process.

Regards,
/Olle



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Re: [asterisk-users] Discussion: Test platform

[Olle E. Johansson]

> 
> Here is the thing, there is nothing stopping 'the community' today from doing 
> this.  In fact, we already have a testsuite [1] in place, running each 
> subversion commit and producing results for the last year.  But this is only 
> one type of testing; automated, we also have unit tests built into Asterisk 
> that run too (EG: a unit test to parse SIP URI). Again, each subversion 
> commit we run the tests and validate results.

I think we should make it more clear and give examples on how we can extend the 
test platform to test functionality in our own platforms - our dialplans and 
channel drivers. If we did that, more people would use the test toolkit and 
work with it daily.

> 
> There is still lots of work that needs to be done though. More test plans and 
> test cases to be added, more code to be written and libraries added, getting 
> more people involved in testing Asterisk Release Candidates (RCs) or patches 
> on the issue tracker.
> 
> That is the hardest part, getting people involved.  Sure it is easy to say 
> Asterisk is not stable, not production ready or it crashes all the time; fair 
> enough but we have tools in place to help resolve that. Just in this thread 
> alone I don't believe one person has answered the call of Olle to volunteer 
> time to help maintain Asterisk 1.4 (if I am incorrect please speak up, I must 
> have missed your name). Additionally, this almost exact point was raise on 
> the asterisk-dev mailing list in 2009 [1] (a great read BTW, lots of great 
> ideas) however due to the lack of interest it did not go to far.
If you go even further back, Russell and I had a branch where we started some 
early work many, many years ago. We're asterisk-dinosaurs in that respect... I 
am very happy that we now, eons later, have a test toolkit. It's lightyears 
ahead of what we discussed or dreamed of back then. And it has helped a lot in 
catching stuff.

> 
> So how can we fix this?  How can we get more people involded?  What makes 
> projects like FedoraTesting[3] and DebianTesting[4] popular?  How can the 
> Asterisk project reproduce their success?
Give them something that tests their own setup as well as test the Asterisk in 
the core.

> 
> As I've said before, I'm more then willing to help with answering questions 
> about the testsuite or reviewing code that people want to get merged in.  We 
> also have an IRC channel, #asterisk-testing available for people to join, ask 
> question, idle, lurk, etc, or if you want to reply to this thread, feel free. 
>  But get involved! :)

Absolutely - we need people that test the new bugs that  developers invent :-)

/O
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Re: [asterisk-users] Orginate not working well with PSTN lines

[Olle E. Johansson]

5 maj 2011 kl. 16.35 skrev Gilles:

> On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell 
> wrote:
>> I know this thread is dead but: I do not believe this should go into the 
>> DAHDI
>> kernel modules.
> 
> I agree. It's just too bad Dahdi is unable to report how an outgoing
> call is doing: Still ringing, busy, answered.
> 
Just to add to the confusion... I have a branch where I managed to get manager 
originate to handle early media.
If we get 183 (sip) or progress in ISDN with media before the answer, a manager 
originate will start the bridge.

We're using that to get the Telco messages when we dial out to connect to a 
meetme. Previously we just had failed calls, but now we can hear the Telco 
message saying something like "Invalid number, please try again" or "Weasles 
have eaten your phone system"

In the SIP channel, I would like to send some sort of control message when we 
get 100 trying. This means that we at least have a connection to something, 
even if we don't know if we've reached the target endpoint.  I don't know if 
there's a similar message in ISDN, PSTN or other channels. 

But that's another patch :-)

/O
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Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

[Olle E. Johansson]

5 maj 2011 kl. 15.11 skrev Paul Hayes:

> On 05/05/11 14:04, Jonas Kellens wrote:
>> Hello list,
>> 
>> 
>> what does this mean :
>> 
>> [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered
>> elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause
>> code, buddy. The cause code!!!
>> [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call was answered
> [snip]
> 
> see rfc3326 section 3.1. "Call Completed Elsewhere".
> 
> It's used so that phones in ring/hunt groups don't record a missed call if 
> the call is answered by someone else.
> 
> I was looking forward to Asterisk supporting this for a while :)
> 
We've had that for quite some time. There's an option to Dial() and one for 
Queue() to enable it. Check the documentation.

/O


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Re: [asterisk-users] SIP secruity: username and password

[Olle E. Johansson]

5 maj 2011 kl. 14.17 skrev Alex Balashov:

> Bilal,
> 
> On 05/05/2011 08:08 AM, bilal ghayyad wrote:
> 
>> When the endpoint register on Asterisk or initiate a call, so they
>> exchange the sip username and password. What is the possibility that
>> this will be capture by the hacker and how to avoid this problem?
> 
> Strictly speaking, there is no inherent connection between either 
> registration or call initiation on the one hand, and authentication. Both of 
> those scenarios can be performed in an authentication-free fashion.  In fact, 
> in most cases the SIP UAC will first attempt to send both a REGISTER and an 
> INVITE request without any authentication credentials.
Because they HAVE TO. In the 401/407 reply, there's a challenge (nonce) which 
is an important part of the MD5 Digest Auth scheme.
> 
> However, it is typical of a SIP UAS providing retail services to the public 
> at large to reply to those requests with a 401 or 407 proxy challenge 
> requesting authentication.  The UAC then resends the request with digest 
> authentication headers, including a password encrypted via a cryptographic 
> one-way hash function.  The entire mechanism was borrowed from HTTP digest 
> authentication.
> 
> The authorisation username can absolutely be intercepted, as it is 
> transmitted it in plain text.  But this is not news.  The password is 
> encrypted, and while the encrypted version can be intercepted, it is 
> encrypted using a one-time "nonce" value that is part of the 401 or 407 
> challenge sent by the UAS.  Nonce values typically have fairly stringent 
> expiration times, at least on good implementations, but nonce replay attacks 
> are possible in principle.
The password is NOT encrypted. It's is used as the basis of a textstring you 
calculate a hash from. That's very different :-)
> 
> This mechanism is reasonably secure, as a compromise with the 
> interoperability requirements of providing SIP service across the public 
> Internet.  In high-stakes situations, however, it may not be sufficient, and 
> may call for SIP over a TLS transport, or encrypted tunnels.
I would say it may call for SIP with TLS client authentication - regardless if 
you need encryption or not...

Cheers,
/O
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Re: [asterisk-users] SIP secruity: username and password

[Olle E. Johansson]

5 maj 2011 kl. 14.08 skrev bilal ghayyad:

> Hi All;
> 
> When the endpoint register on Asterisk or initiate a call, so they exchange 
> the sip username and password. What is the possibility that this will be 
> capture by the hacker and how to avoid this problem?

We never exchange passwords in clear text in SIP 2.0. SIP uses HTTP digest 
authentication with MD5. There are many articles about that on the web, so that 
you can find out how it works and what the risks are.

Cheers,
/O
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

5 maj 2011 kl. 12.04 skrev Paul Hayes:

> On 05/05/11 05:41, Cary Fitch wrote:
>> 
>> 
>>Flavio E. Goncalves
>>www.asteriskguide.com 
>> 
>>Compare to which version of Windows… Patches are a never ending process
>> 
>>Cary Fitch
>> 
>> 
> 
> I think this attitude is half the problem.  Asterisk is not a desktop 
> computer operating system.  It is the engine for a telephone system, a 
> telephone system needs to be much more reliable than a desktop PC if it is 
> going to continue to compete in a growing industry.
> 
> I agree with the comments on concentrating more on stability than new 
> features.  It's hard because it is new features that make good stories and 
> are easier to shout about in order to get a product better known.
> 
> For now I am sticking with 1.4 mainly (although I am using 1.6 where I need 
> BRI connectivity) but my plan is to move to 1.8 when I feel I have tested it 
> enough and it's been around for long enough to be proven.

Great. You are part of the test team :-)

One has to remember that this is open source. We need to work together to 
stabilize 1.8. No one else is going to do it for it - and I feel it needs to be 
done.

Thanks for your help!

/O


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

5 maj 2011 kl. 06.33 skrev Olivier:

> 
> 
> 2011/5/5 Flavio Goncalves 
> 
> but stuffing Asterisk with
> many  new features on each version does not seem to be contributing to
> the stability of the code or the migration to newer versions.
> 
> yes but it seems to me that code stability is improving.
> Maybe next 1.10.0 version will be "production-ready" from day 1 ?

Unless a lot of users step in to test the pre-releases, that will not happen 
with new code in this project or any other project. It just takes time. The 
more people that test, file bug reports, patch code and helps us through the 
process, the better.

I would like to suggest that the community put more eyes towards helping with 
the test system. The test system, as Russell pointed out earlier, is a huge 
improvement that saves us from repeating a lot of mistakes. Ideally, for every 
bug fix we should add a test to make sure it doesn't come back... The test 
system makes every release better than previous releases. And it can help in 
your installation as well!

/O




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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

5 maj 2011 kl. 05.28 skrev Flavio Goncalves:

> My 2 cents. All these problems seem to be lack of focus. Digium,
> please stop doing everything to everyone. Too many versions, too many
> features, too many code, too many bugs. Following the Pareto's
> principle, 80% of the users use only 20% of the code. My suggestion is
> to start thinking of Asterisk as a platform taking care of only 20% of
> the code. Digium is in position to create a market place for free and
> commercial Asterisk applications, drivers and modules. Look at some
> other open source communities such as Joomla at
> http://extensions.joomla.org, There are more than a thousand modules
> maintained by the community. Imagine, do you want a multitenant
> parking module? Great there is one in Digium App Store for a few
> dollars. Digium could have its own commercial modules. Support for 3rd
> party applications would be up to the 3rd party developers. Why iPhone
> developers make money and Asterisk developer's usually don't? If
> people pay for silly games in iPhones wouldn't they pay for a Unistim
> driver if they have hundreds of compatible phones?

What you are describing is in the architecture for Asterisk SCF. The "F" stands 
for Framework, which hints at the ideas behind it.

Asterisk as it exists today has been around for a long time. There are many, 
many extensions around and we also have the forge and the marketplace. It will 
be hard removing stuff from the distribution, imagine the reaction - 
considering my latest reaction to the 1.4 actions :-)

> 
> I would like to say that I have a deep respect for Asterisk and Digium
> that redefined the global telephony market, but stuffing Asterisk with
> many  new features on each version does not seem to be contributing to
> the stability of the code or the migration to newer versions.

Thanks for your input!

/O
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Re: [asterisk-users] Password to be ecrypted?

[Olle E. Johansson]

4 maj 2011 kl. 19.44 skrev Robles Román, José Miguel:

> By the way, I like the implementation in iax.conf (auth=md5 ... 
> secret=x), it seems more flexible, and it enables the use of other hash 
> functions or other security algorithms.

The SIP protocol does not support any other hash functions today.

/O
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Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

[Olle E. Johansson]

2 maj 2011 kl. 18.09 skrev Hans Witvliet:

> On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote:
>> Hi everyone,
>> 
>> 
>> How can I introduce some distortion, echo, chopping sound and all
>> other bad quality things that can happen to a SIP trunk? I have plenty
>> of bandwidth and crisp clear lines so the only thing that I can think
>> of is to limit bandwidth but even that requires quite some scripting
>> work. 
>> 
>> 
>> Is there any easy way to simulate a distorted SIP line temporarily for
>> testing?
> 
> You can intruduce a predefined amount of "distortion" on your ip-connection
> (packet loss, fluctuating delay, out of secuence reception of packets,
> limited bandwith)
> 
> All of these will have a serious impact on your VOIP-connection.
> 
> See "lartc" about it.
> Good thing about it, is that you pre-define how bad a line is, and it
> produces re-producable results

I use a laptop with a usb-ethernet connected in bridge mode as a "voip 
destroyer".
Using TC you can inject a lot of bad stuff on the connection.

/O
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Re: [asterisk-users] Discussion: 1.8 quality issues

[Olle E. Johansson]

28 apr 2011 kl. 16.53 skrev Russell Bryant:

> 
> - Original Message -
>> PS. Please don't start a discussion about 1.8 quality in this thread,
>> that's a separate issue. I just want to know what you think about
>> closing 1.4 support now. If you want to discuss 1.8 quality, start a
>> new thread. Thanks.
> 
> I don't think it's a separate issue at all.  I would like to see discussion 
> of exactly which issues are preventing users from using Asterisk 1.8.  We're 
> trying to shift focus to those issues and get them resolved as quickly and as 
> efficiently as we can so that we can all move forward.
Thanks for ignoring my plea... Please at least change the subject ;-)
> 
> Resources are limited.  What is the best use of our time to help ensure the 
> best future?  Where do we want to see the project in the next 6 months to a 
> year?  A primary focus on further solidifying Asterisk 1.8 is what gets us 
> there in my mind.
I agree.
> 
> Asterisk 1.4 was released 4.5 years ago.  It mostly "just works", and I fully 
> expect many to keep using it until they see a need to migrate.  
If you think it's mostly "just works" it can't be hard to support it a while 
longer then, can it?

> This process has been likened to when the community moved from Asterisk 1.2 
> to 1.4.  Asterisk 1.8 has been much more stable out of the gate than 1.4, due 
> to many things we have done over the years to increase quality, including:
> 
> 1) We have adopted peer code reviews as common practice for all non-trivial 
> changes going into Asterisk.  This alone has _greatly_ increased the quality 
> of the code going in.  It is rare that a patch goes up for review where 
> someone doesn't point out some sort of problem.  These problems are found and 
> fixed _much_ faster in the up front review process than if it had been many 
> months later when someone encountered it as a bug in the field.
Agree. But it also puts a significant delay on the process. We have to be very 
careful about that. Having too many branches open in addition to this was a 
pain. With fewer branches I hope it will get better.

> 
> 2) We have placed an increased emphasis on automated testing efforts.  In 
> addition to building up a lot of test environments inside of Digium, there is 
> now an open source automated testing effort for Asterisk.  There are over 200 
> test cases that run every time anyone touches the code.  This includes 
> complex call scenarios such as transfers and call parking.  These open source 
> test cases touch about 25% of the code (and what it does touch are things we 
> considered some of the most important parts).  That is a huge step forward 
> from where we started.  We are continuing to place more and more resources on 
> this effort to move it forward.
Agree. It's great and we need to continue working on it, because it obviously 
hasn't caught everything we should have caught. I fully agree that it is a 
wonderful system and I've said that many, many times.
> 
> Despite comments in this thread, there _are_ many people using Asterisk 1.8 
> in production, including large installations.  The ones with systems working 
> perfectly fine don't tend to make as much noise.  :-)  For those still 
> getting hit by problems, I hope that you can make the time to report them so 
> that we can work with you to get them resolved.
I don't disagree there either. I have only stated that it fails in my and my 
customer's installations. Everyone is using Asterisk in different ways. If it 
did not work anywhere I would be very disappointed.
> 
> I started my work on Asterisk as a volunteer 7 years ago and even though it 
> is now my full time job, I still put many personal hours into the project.  I 
> care very deeply about the success of Asterisk.  I truly believe that the 
> steps we have taken with release management are in the best interest of the 
> project.
I understand that you do, I don't think you do things you don't believe in. But 
you do need feedback from production sites to make the best decisions. 

What you bring up here is important but in my world have no relation to the 
decision about 1.4. I understand you want to use development resources in a 
good way, but there are also marketing/business perspectives to consider here. 
I personally don't think closing 1.4 support today is in the best interest of 
the project from a marketing point of view, as I don't believe we have a 
working alternative to offer. I understand we have different opinions about it.

Regards,
/Olle
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

29 apr 2011 kl. 01.49 skrev Leif Madsen:

> Well the issue is that we currently have over 900 open issues in the Asterisk
> project alone, and with only one primary bug marshal (myself) sometimes things
> accidentally get closed if it looks like a configuration issue.

What's the reason that we only have one bug marshal? We used to ask people to 
become bug marshals to help,
but the last I heard you and Russell did not want community marshals. What went 
wrong with that? Wasn't it any help?

/O
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

>> 
> 
> I don't think there's anything inherently wrong with the bug tracking system. 
>  It's more of a resource issue with many conflicting priorities.  Officially 
> letting off some of the pressure from older branches does help.  I would like 
> to be making faster progress through bug reports and patches.  I do have an 
> open position for another full time Asterisk developer at Digium in case 
> anyone is interested.  :-)
> 

I agree with Russell here, we have resource issues in the bug tracker but 
that's nothing that can be solved by another piece of software. If you have 
issues that is not handled timely, why don't you spend some time with other 
issues to help out? Surely there are issues where you can give a helping hand.

In answer to an earlier email that I felt was kind of attacking me I would like 
to point out that I am very happy and grateful about the resources that Digium 
put in the project, and continue to do. Just to clarify that this discussion 
was not about trying to paint Digium as a company as evil - which I was accused 
of. Digium is a very old business partner to my company and we've done great 
together. That doesn't mean we can't critizise each other or not want to 
discuss issues in the open.

To answer another attack, I have been contributing code and bug fixes to both 
1.8 and trunk. Most of my code exist in versions for trunk and 1.4. Customers 
pay me for 1.4, I forward port it to trunk when I have time and resources over. 
It's not a personal choice that most of my development work still is based on 
1.4. Of course I would love being doing development freely, creating great new 
code for the new release. There's a lot of stuff to do in Asterisk trunk, but 
no one out there that wants to put resources towards it in my direction. 
Asterisk trunk development is sadly too far away from my customers current 
business. The 1.6.x release schedule widened that gap and we need to discuss 
how to close the gap again. We do not need a large number of maintained 
releases between the long term support releases.

So far I haven't seen more than a few people that chimes in to this discussion 
saying we need to have 1.4 open, I haven't seen many people running 1.8 in 
production either. I have seen a lot of important issues being reported with 
1.8 which to me confirms that it's still not ready.

I have been working in commercial software companies for a long period in my 
life. A product manager that called for end-of-life of the 1.4 release at this 
stage would be out of a job very soon. Migrating a customer base from one 
version to another is very, very hard. It seems much harder in telecom software 
than in the rest of the software world. We need to continue to work on 1.8 and 
do a lot of marketing for upgrading as soon as we're comfortable with it and 
have resources to manage the bug reports that will come in. We really need to 
push and shove. What I can't do to my customers is forcing them to upgrade to 
something that doesn't work. Customer will simply stop paying me if I do.

I will not continue to push this issue, just realize that I will have to manage 
my own 1.4 branch fixing the issues that affect my customers, which will 
exclude management of a lot of modules that are not used at all in our 
installations. As I said before, I have no resources to support all of the code 
base for everyone. That's just life, painful as it is. In the ideal world, 
there would be resources to help everyone. Unfortunately, I still have to have 
money to bring home at the end of the day.

Thanks for a very good discussion. As usual, I learned a lot from it. Keep 
reporting issues so that all of us can move forward to new releases.

Feel free to contact me off-list if you want to discuss this further.

/O


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

> 
> I(me, my opinion, my feelings, my commercial view) am on the side of
> dropping support for 1.4 and 1.6.  1.8 had some major issues which are
> resolved/being worked on with more energy as older platforms are shut
> down. If a large enough security issue showed up, I hope we would all
> try to do the right thing and push it back to 1.6 and 1.4.
1.6.x is not an option for me at all. These' releases are not LTS. We can't 
upgrade as often as that release schedule required. I am very happy to see 
1.6.x disappear
in the darkness and from my hard disk drives.

> Support
> must end sometime. Merging changes across many versions is very
> difficult and time consuming.  
I fully agree here.

> Asterisk GUI is very limited do to its
> 1.4 support code.  There are users that still use 1.2 and are very
> happy.  They are not looking for new features. I hope the 1.4 / 1.6
> users can survive while they test the 1.8 branch and share why or why
> not it will fit their needs.

They will survive and they will merge their own bug fixes. I just wish we could 
share the work and maintain the branch in public instead of everyone managing 
it by their own. As long as 1.8 is not ready for the way we use it, we have no 
version to migrate to. 

I am sure that 1.8 will fit their needs and deliver a lot of extra. It's a cool 
new release. Everyone wants to go there. That's not the issue here. The issue 
is when it's ready for the larger installed base beyond the early adoptors.

I don't like the project I've been part of for many years not offering a 
supported option that fits the customers I work with. It's as simple as that. 

Saying that they should know better, that the project has posted the release 
plans for a long time warning about this - it  just doesn't cut it as long as 
we have no working code to replace the current version with. 

Compared with last time we had a painful migration (from 1.2 to 1.4) there are 
numerous other options out there.  I think the project have to be a bit more 
careful about our attitude towards the installed base. I want to keep them in 
the Asterisk project. That is where I belong and where they belong.

/O
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[asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Olle E. Johansson]

Friends,

We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
According to the release plans, support for 1.4 was scheduled to close in April 
2011 - basically now. After that, only security patches would be committed. 
This is already a delay from the original plan published by Russell Bryant.

Unfortunately, I think this is way too early. My feeling and experience is that 
1.8 is not ready for production in the environments I work in - large scale 
installations. Customers are not planning migration and all new installs are 
still 1.4. Tests we've been doing with 1.8 has failed within just a short time 
and so badly that customers has not paid me to spend any further time with 1.8.

Last time we went through this process with a LTS release (which we did not 
know then) it took over one year before we had a stable product to migrate away 
from 1.2 and jump on the 1.4 track. Hopefully, with the help of community, we 
can move up to 1.8 late this year or early next year. For me 1.8 is the focus, 
it's the LTS release.

Not having a supported 1.4 version from the Digium-hosted repositories will 
mean that we will have to move to separate repositories or branch off from the 
main track. I already maintain a ton of subversion branches with various 
patches to 1.4 It takes a lot of time to manage this version that is a fork 
from the main 1.4 branch. I will soon have to start working with subversion 
branches for 1.8 to create a compatible version for my customers to test, since 
most of the patches is not part of 1.8. After a few years of doing this, I know 
the work involved with managing code myself.

The Digium team wants to go ahead and not support 1.4 any more, I want to keep 
1.4 open for normal bug fixes. What do you think?

Kevin proposed that the community maintains the 1.4 branch without support from 
the Digium team. I don't think that's a good solution, but it may be the only 
solution.  I haven't got the resources to manage the 1.4 code myself, so I 
won't step forward as a maintainer if I can't get proper funding. Anyone else 
out there that has the time and resources to manage the code?

Feel free to send me mail off list if you have ideas or suggestions on how to 
solve this - or continue the discussion here.

Regards,
/Olle

PS. Please don't start a discussion about 1.8 quality in this thread, that's a 
separate issue. I just want to know what you think about closing 1.4 support 
now. If you want to discuss 1.8 quality, start a new thread. Thanks.
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[asterisk-users] The SIP channel driver - I'm giving up.

[Olle E. Johansson]

Friends,

After having spent many years working with the Asterisk SIP channel driver and 
the SIPv2 protocol, I have finally realized that this is a dead end. It's 
getting nowhere and it's way too complicated to set up, run and support in 
working code.

After realizing this, I started a new standardization project together with my 
friends in Canada, Simon and Marc, to develop a working solution based on the 
combination of IPv6 and SIP. We have gotten great feedback and now the IETF, 
the ITU and the IPv6 forum jointly launches the new standard, SIP-six.

>From the press release:

"”We realize that 99% of the SIP users use SIP for PSTN phone calls. The 
original SIP standards was written with other applications in mind, a vision 
that never came true.” said Bob Plug, area director in the IETF. ”That’s why we 
sat down and said to ourselves that this should be way more simple.”

The SIP-six standard totally removes the dependency of domains and URI syntax. 
There’s no point in using this, since everyone seems to think that IP 
addressing is more than enough. The new standard use part of the vast IPv6 
address space to incorporate the E.164 phone numbers as addresses. This is the 
reverse of the reverse phone number usage in the enum standard, which is no 
longer needed in SIP-six.

By using IPv6 mobile IP, phone users register their phones and get access to 
their phone number. Users that need security can easily integrate IPsec into 
their setup. Media and signalling uses the same addressing scheme and is mixed 
so that both SIP-six, RTP and RTCP only uses one port address - but in this 
case a port address with 32 bit subaddress identifying the media stream. This 
finally solves a lot of the firewall traversal issues that SIP v2.0 had. With 
the combination of mobile IP and use of public IPv6 addresses NAT traversal 
won’t be an issue.

The testbed for SIP-six has been running for a year at six choosen large SIP 
carriers, with the assistance of Edvina AB in Sweden and ViaGenius in Montreal, 
Canada. In an International effort, the testbed is today put in production and 
Roboid phones all over the world is automatically connected to this worldwide 
network."


You will be able to find out more about it here: 
 http://bit.ly/sipsix

SIP-six is implemented as a channel driver in Asterisk 2.0, as a replacement 
for SIP2.0 in Kamailio 4.0 and a channel module in FreeSwitch - all releases to 
be released later today. Softphones for testing will shortly be available from 
Blink and Zoiper.

Looking forward to continue this project with the rest of the Asterisk 
community!

Have a nice weekend!

/Olle
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Re: [asterisk-users] Multiple SIP endpoint registrations

[Olle E. Johansson]

16 mar 2011 kl. 14.13 skrev Benny Amorsen:

> "Kevin P. Fleming"  writes:
> 
>> Why do you need a Local channel to do this? If extension 234 exists in
>> some context, the Dial() statement in that extension can dial
>> SIP/234-foo and SIP/234-bar itself.
> 
> Good point.
> 
> It can be a bit of fun keeping track of the phones when they are
> added to or removed from queues, and the owner expects both of them to be
> added/removed at the same time. It is still doable without Local channels.
> 
> Once you need to do manipulation of calls before passing them on to the
> phone (change callerid individually, handle tT options etc.), Local is
> unavoidable, but at that point multiple registrations would not work
> either.

We also need to adopt SIP headers depending on device - so we need a dial per 
device, not one combined.

/O
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Re: [asterisk-users] Realtime SIP, multiple AX servers question

[Olle E. Johansson]

3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot:

> 
> Normally, no matter which Asterisk server an ATA connects to, we get our 
> database fields filled out correctly, such as "regseconds", "lastms", 
> "ipadr", etc. However, with some ATA's we are experiencing a problem as 
> follows:
> 
> 1. ATA reaches its "re-registration" timeout, which we typically configure to 
> be 60 minutes.
> 2. ATA re-queries DNS SRV record, and ends up re-registering with a different 
> AX server than it was on previously.
> 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc).
> 4. The old AX server, after a few more minutes, notices that the ATA has 
> vanished, and therefore clears out these same fields.

Oh, that's an interesting observation. Depending on how you see it, it's a bug 
or a feature request.

Code-wise what you could do is that Asterisk could retrieve the information 
from realtime. If the sysname is not the same as the systems, it let the 
information be. If it's the same sysname, then erase the information.

/O
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Re: [asterisk-users] changing sip port

[Olle E. Johansson]

11 nov 2010 kl. 23.25 skrev Baha @ SH:

> Hello
> How can I run the sip service on asterisk on another port beside 5080?
> I mean asterisk will still take sip requests on port:5080 and another custom 
> port, lets say port:6080
For UDP, we only have one port. You have to select.

/O


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Re: [asterisk-users] OT: certificate for softphone

[Olle E. Johansson]

10 nov 2010 kl. 21.48 skrev Hans Witvliet:

> On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote:
>> 6 nov 2010 kl. 15.30 skrev Hans Witvliet:
>> 
>>> Hi all,
>>> 
>>> As stated in the subject, slightly off-topic, as it is not directly a
>>> Asterisk issue, but more SIP in general
>>> 
>>> Because security in general, and specifically identification becomes
>>> more and more a subject for more concern, and Asterisk is capable of
>>> doing sip/TLS, i was wondering what more could be done to improve
>>> security.
>>> 
>>> Specially softphones, might it be possible to employ etokens or
>>> smartcards for holding the certificates needed by TLS?
>>> 
>>> Done before?
>> 
>> In the SIP protocol there is support for TLS client certificates, much like 
>> in HTTP. 
>> 
>> Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in 
>> front of Asterisk to get this kind of strong authentication.
>> 
>> /O
> Am i that mistaken?
> 
> I got the impression** that sip-registration of a phone could be done in
> the same way as client-authentication on apache:
> On the server-side you got the certificate holding your public key which
> is signed by a trusted third party (the CA), while you hold your private
> key on a smartcard or token. If you start your browser you are prompted
> for your pin-code.
> 
> I was just hoping that there would be a softphone that could work the
> same way, two-factor authentication.
> 
I haven't seen any soft clients implementing this. Bria/Eyebeam may have it, 
but they've removed all TLS options from the GUI.

As I said, the SIP protocol supports it. Kamailio supports it on the server 
side. Now we need clients that supports it.

Now we're talking about authentication. For identity assurance, there's another 
set of standards called SIP Identity where you use TLS to sign your identity.
The TLS is just between the phone and the first server. Identity is supposed to 
be something that follows the call to the callee.

/O


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Re: [asterisk-users] Feature Request for 1.10 - ISDN power-save mode

[Olle E. Johansson]

2 nov 2010 kl. 17.19 skrev Olivier:

> Hi,
> 
> In Europe many Telcos implement power-save mode
> (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
> 'Activation / Deactivation' for more information).
> 
> Would you agree to have this feature added to the ones already discuused for 
> next Asterisk release ?
> (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010)
> 
The projects you see on that list all have resources allocated to them or 
reasonable close to get allocated by the persons that participated in that 
meeting - unless you find them in the final categories (3.9 and 3.10).

If you have development resources or funding and can create code that works, we 
are ALWAYS open for contributions, regardless of our lists.

Looking forward to your contribution!

Best regards,
/Olle
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Re: [asterisk-users] Exceptionally long queue length queuing . . . .

[Olle E. Johansson]

31 okt 2010 kl. 13.43 skrev Paul Belanger:

> On Sat, Oct 30, 2010 at 6:22 PM, Brian Capouch  wrote:
>> I wonder if anyone out there has a perspective on this.  There are a
>> welter of tickets out there on the matter, most of them closed.
>> 
> I'm actually able to reproduce this pretty often, for me using IAX2
> with IMAP voicemail (google apps) is how.  I haven't had much time to
> debug it, but plan to play more with it the coming weeks.

Any update, Paul?

/O

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Re: [asterisk-users] OT: certificate for softphone

[Olle E. Johansson]

6 nov 2010 kl. 15.30 skrev Hans Witvliet:

> Hi all,
> 
> As stated in the subject, slightly off-topic, as it is not directly a
> Asterisk issue, but more SIP in general
> 
> Because security in general, and specifically identification becomes
> more and more a subject for more concern, and Asterisk is capable of
> doing sip/TLS, i was wondering what more could be done to improve
> security.
> 
> Specially softphones, might it be possible to employ etokens or
> smartcards for holding the certificates needed by TLS?
> 
> Done before?

In the SIP protocol there is support for TLS client certificates, much like in 
HTTP. 

Asterisk doesn't support it. You need to put a SIP proxy like Kamailio in front 
of Asterisk to get this kind of strong authentication.

/O
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem

[Olle E. Johansson]

10 nov 2010 kl. 02.38 skrev Brett Woollum:

> Good idea Paul.
> 
> My debug output:
> [Nov  9 17:33:39] VERBOSE[2923] netsock.c:   == Using SIP RTP CoS mark 5
> [Nov  9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1] 
> Set("SIP/413-0005", "CALLERID(num)=2") in new stack
> [Nov  9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2] 
> NoOp("SIP/413-0005", "CallerID(num) is: 2") in new stack
> [Nov  9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3] 
> Dial("SIP/413-0005", "SIP/412") in new stack
> [Nov  9 17:33:39] VERBOSE[4175] netsock.c:   == Using SIP RTP CoS mark 5
> [Nov  9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
> [Nov  9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
> [Nov  9 17:33:44] VERBOSE[4175] pbx.c:   == Spawn extension (sipphones, 412, 
> 3) exited non-zero on 'SIP/413-0005'
> [Nov  9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1] 
> Hangup("SIP/413-0005", "") in new stack
> [Nov  9 17:33:44] VERBOSE[4175] pbx.c:   == Spawn extension (sipphones, h, 1) 
> exited non-zero on 'SIP/413-0005'
> 
> As you can see on line 3, CallerID(num) was successfully set to "2". 
> SIP/412 is dialed next. It receives the call, but with "412" as the Caller ID 
> number - even though the real source of the call was extension 413. The name 
> I set in CallerID(name) works fine. 
> 
> My Extensions.conf for that context:
> [sipphones]
> exten => 412,1,Set(CALLERID(num)=2)
> exten => 412,1,Set(CALLERID(all)="TEST"<2>)
> exten => 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
> exten => 412,n,Dial(SIP/412)
> exten => 412,n,NoOp(${CALLERID(num)})
> 
> If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 
> into sip.conf directly, this code works (ie: the CallerID(num) I set makes it 
> out to the destination phone properly).
Have you set the fromuser= field in the realtime database?

/O
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[asterisk-users] Asterisk in the third world - Astricon 2010 keynote follow-up

[Olle E. Johansson]

Friends,
After listening to Mark Summer's keynote at Astricon (hopefully soon on the 
Astricon web site) I think we should come back to the discussion he started. 
Mark talked about using Open Source in general and Asterisk in particular in 
third world projects as well as in emergencies in other countries. He and 
Inveneo help groups of people to get a better understanding of how to build 
network, IP and voice infrastructures. One part is of course learning and 
managing Asterisk.

I do believe many of us wants to help his efforts, but lack the understanding 
and channels to reach out. I had a very brief discussion with Mark after the 
keynote and promised to get back to him.

My thoughts are that if anyone from these countries try to reach us, we fail to 
listen and help. Could be language, could be attitude or something else. We 
can't expect them to have full understanding of net etiquette, the rules of 
Open Source project management or how to find information themselves (in a 
language they might not understand fully). The climate in our mailing lists and 
chat rooms are not always one of understanding, especially if someone copies 
their english language and attitude from Miami Vice ;-)

Do you have any ideas of what could be done from our community? Can we create 
special forums where we have a different climate, more languages and better 
understanding?

I also think we should copy ISOCs efforts and have a pre-astricon 
training/workshop for people that Inveneo locate and then invite them to 
Astricon, funded by grants form community or from somewhere else (since we lack 
an "Asterisk foundation" that could help here). I'm sure we can find resources 
to get them to Astricon and that we can find teachers in the community that are 
willing to help with this project. I would not hesitate in donating a few days 
myself.

We have enormous powers in our community. If we can gather a small part of that 
and point it towards these people, we can change the situation for many more, 
just by doing what we do each day - enjoy building voice solutions and sharing 
our knowledge.


Let's brainstorm for a while!  The floor is open.

/O
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Re: [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais

[Olle E. Johansson]

11 aug 2010 kl. 15.49 skrev Leif Madsen:

> On 10-08-10 04:11 AM, Olle E. Johansson wrote:
>> 
>> 26 jul 2010 kl. 18.13 skrev Leif Madsen:
>> 
>>> On Asterisk 1.6.2, your only option for distributing device state is with
>>> res_ais. I've used it in a labbing system and it works well -- the caveat is
>>> that your machines need to be on a low latency network (i.e. LAN).
>>> 
>>> With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute 
>>> your
>>> device states over the WAN. I've made it work with the Tigase XMPP server. 
>>> More
>>> information about it can be found in the doc/distributed_devstate-XMPP.txt 
>>> file.
>>> 
>> This fall, we're going to implement it using SIP for 1.4 and 1.8/trunk.
>> Look for project "pinana". Development will start later this month.
> 
> Sounds very cool! I look forward to playing around with it. Also thanks for 
> picking a branch name that is not related to fruit or frogs.

Thanks for the feedback. I guess the name was a mistake and I'll take it under
reconsideration :-)

/O
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Re: [asterisk-users] asterisk & distributed device state => res_jabber Versus res_ais

[Olle E. Johansson]

26 jul 2010 kl. 18.13 skrev Leif Madsen:

> On 10-07-26 10:45 AM, Mathieu wrote:
>> Hello,
>>  as I'm looking for a solution (with asterisk 1.6.2) , my
>> investigations leaded to :
>> - res_ais =>  libais&  corosync. (each node need to run corosync / aiexec)
>> - res_jabber =>  libjabber&  iksemel. (each node need to be connected on
>> an XMPP server)
>> 
>> I've been able to make some successful tests with res_ais on 2 servers
>> but got some CPU issues with corosync after some hours of activity.
>> 
>> What's the best solution regarding flexibility and stability and
>> real-time exploitation ?
>> 
>> I've got the feeling a good (and old) XMPP server will be more reliable
>> than res_ais which seems to be pretty young.
> 
> On Asterisk 1.6.2, your only option for distributing device state is with 
> res_ais. I've used it in a labbing system and it works well -- the caveat is 
> that your machines need to be on a low latency network (i.e. LAN).
> 
> With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your 
> device states over the WAN. I've made it work with the Tigase XMPP server. 
> More 
> information about it can be found in the doc/distributed_devstate-XMPP.txt 
> file.
> 
This fall, we're going to implement it using SIP for 1.4 and 1.8/trunk. 
Look for project "pinana". Development will start later this month.

/O


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Re: [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]

[Olle E. Johansson]

> 
> Further to Steve Edward's comment, I think things would be more
> obvious if the help system was improved slightly, for instance:
> 
> If you were trying to figure out the commands dealing with peers, you
> would be able to type:
> *CLI> help peer
> No "peer" command found.  Possible alternatives:
>iax2 show peer Show details on specific IAX peer
>   iax2 show peers List defined IAX peers
>sip show peers List defined SIP peers
> sip show peer Show details on specific SIP peer
>  (and so on, maybe using the "[More]" option to help it be readable)
> 
> In this case, if I could use the "help" system to search on all
> occurrences of the word "hangup" in the available commands, I would
> probably have found it myself instead of bothering the list.

THat's a very good idea. Thank you! 

Now we need someone that codes it :-)

/O
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[asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8

[Olle E. Johansson]

an Danny Wyndham and the 
Swenglish dialect of Asterisk developer and guru Olle E. Johansson, one that
was recognized with a strange smile by all Asterisk developers testing VCC.

VCCnet technology includes scalability and security components  licensed by
Edvina AB in Sweden. Edvina's experience of large scale Unified Communication
networks was necessary to build a world-wide network-centric platform for 
this new service. 

- "We find it exciting to contribute to this new service. Realizing the perfect
match between the open IPv6 protocol and the proprietary Dundi technology
was an eye-opener. No NAT issues and the possibility to build a worldwide
network with service discovery, security and managed QoS will make this
a success story. We're proud to contribute to this solution." says Olle E. 
Johansson,
founder of Edvina. "The new IAX3 protocol is also really interesting, as it
not only combines media and signalling over one port, but now also adds
presence, instant messaging, file transfer, printing, database queries, 
directory
services and network management  over the same port. It's a one-size-fits-all 
protocol that will handle all services a user want."

The VCCnet network is already in operation, The VCCnet PBX interface will be 
part
of Asterisk 1.8 to be launched later this year and part of the VoxSwitch update
Q2 2010. The VCCstore opens June 1st. Development kits are available to
Digium authorized VCC development partners today. The VCC technology
is patented by Digium and will be operated as a private virtual network on top
of the Internet and the ISDN network.

For questions and further information, please contact the Digium marketing 
department at
loofli...@digium.com today. A press conference will be held April 1st, 15:00 
GSM+1 in 
VCCconference room 142857 for media representatives. It will be available for 
one 
week on vcc://digium.com::conference:142857 for later viewing.

VCC, VCCnet, VCCblock, VCCstore, Digium, IAX3, Dundi and Asterisk are
trademarks registered by Digium Inc.



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Re: [asterisk-users] rtp.conf ports for inbound or outbound?

[Olle E. Johansson]

25 mar 2010 kl. 13.14 skrev Michelle Dupuis:

> I can't find this in the wiki/email history..but I'm sure it's based asked 
> before.
>  
> The port range define in rtp.conf - is that for connections initiated by 
> asterisk?  Or the port range asterisk listens on?  Or both?
>  

These are the ports Asterisk use for INCOMING media, the ports we listen on.

/O
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Re: [asterisk-users] permit/deny in sip.conf iax.conf

[Olle E. Johansson]

24 mar 2010 kl. 16.48 skrev Karl Fife:

>>> Steve Edwards wrote:
>>> 
 It may not be as intended, but from a "user" standpoint, it seems 
 logical
 and convenient to establish "policy" in [general] and make exceptions in
 the entities as needed.
>>> 
>>> Right... for when you have one policy. When you have two policies, each
>>> that apply to a dozen or more entries in the config file, then it really
>>> doesn't help, it harms. Templates solve that problem completely, because
>>> each policy can be its own (named!) template, and they can be combined.
>>> Since templates are also very easy to use for the single policy case,
>>> they are a better solution to teach people (and they're also easier to
>>> implement in the configuration code of the module).
>>> 
>>> In other modules created since chan_sip, we've intentionally avoided
>>> this problem, and you'll note that in nearly every other module, the
>>> [general] section is exactly that; general settings for the module, and
>>> not defaults.
>> 
>> In my NACL work, I implemented a channel-wide NACL for blacklist purposes.
> 
> Can you talk more about this?  Were your Named ACL's something other than 
> templates?
> 
> What was/were the specific 'pain point/s' you were trying to assuage?  For 
> example did you need something not currently offered in the existing 
> frameworks, for example DNS-resolved hostnames for permitting/restricting 
> registration/connection?  Or were you just doing a 
> clever/elaborate/well-implemented setup of the existing frameworks?
> 
> I for one would love to hear your 10,000 foot concepts and any details you'd 
> be willing to share.
Well, I've written several mails and blog entries about this. Many discussions
about security in Asterisk has ended with the need for a new concept
for ACLs, something that can be manipulated by Asterisk using the C API,
by using manager and the CLI. So currently, it's a framework. You can
create a named ACL that is used by multiple devices or SIP trunks.

In the future, we have the API to build all kind of blacklist/whitelist 
functions.
And I'm open for input on what's needed here. Now we have the framework
to build on.

http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists-asterisk-nacls/
http://svnview.digium.com/svn/asterisk/team/oej/deluxepine-1.4/README.nacl

It's something I'm working on just for fun, so it moves slowly forward.

/O
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Re: [asterisk-users] permit/deny in sip.conf iax.conf

[Olle E. Johansson]

23 mar 2010 kl. 22.20 skrev Kevin P. Fleming:

> Steve Edwards wrote:
> 
>> It may not be as intended, but from a "user" standpoint, it seems logical 
>> and convenient to establish "policy" in [general] and make exceptions in 
>> the entities as needed.
> 
> Right... for when you have one policy. When you have two policies, each
> that apply to a dozen or more entries in the config file, then it really
> doesn't help, it harms. Templates solve that problem completely, because
> each policy can be its own (named!) template, and they can be combined.
> Since templates are also very easy to use for the single policy case,
> they are a better solution to teach people (and they're also easier to
> implement in the configuration code of the module).
> 
> In other modules created since chan_sip, we've intentionally avoided
> this problem, and you'll note that in nearly every other module, the
> [general] section is exactly that; general settings for the module, and
> not defaults.

In my NACL work, I implemented a channel-wide NACL for blacklist purposes.

/O

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[asterisk-users] In Berlin this week? Kamailio/Asterisk community dinner on Thursday

[Olle E. Johansson]

Friends,

Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When 
travelling around like this, we often invite the community to come and meet us 
in a nice restaurant. We offer good company and fun discussions about Kamailio, 
SIP-router.org and Asterisk - but the drinks and food are on you. At least 
yours :-) 

Berlin is the city where Sip Express Router was born. Many SER/SIP-router and 
Kamailio developers live here, so we suspect that you'll find a good set of 
core developers joining us.

Hint: Buying a beer for a developer is generally considered a good thing. 
Buying too many will affect the commits the next day... The bad code 
submissions can be reverted easily, so don't worry about it. We'll just have to 
handle the situation...

- Where?  The Lemke Brauhaus, Luisenplatz1, 10585 Berlin (close to Schloss 
Charlottenburg).
- Time? 19.00 Berlin time
- URL: http://www.brauhaus-lemke.com/index.php?area=4

Please send me a not off-list if you think you can participate, so that we can 
get a properly sized table. If you want to take a chance, just show up. Either 
way, you're welcome!

This is also a good way to prepare for the VoipAthon - the 24 hour Voip Users 
Group session. Don't miss that!
http://voipathon.org/

The next Asterisk SIP Masterclass will be hosted by Telespeak in the UK. Check 
their web site for information!
I suspect we can find beer or someting compatible in that area too :-)

Regards,
/Olle


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Re: [asterisk-users] SIP codec negotiation / manipulation

[Olle E. Johansson]

22 mar 2010 kl. 14.54 skrev Kevin Sandy:

> 
> 
> On 3/21/2010 4:05 AM, Olle E. Johansson wrote:
>> 
> 
>> 17 mar 2010 kl. 16.37 skrev Kevin Sandy:
>> 
>>> We're having an odd issue with codec negotiation from one of our
>>> SIP providers. Here's the basic situation.
>>> 
>>> We receive an invite from them advertising support for G711, G729,
>>> and G723. In our response, we send back that we support G711 and
>>> G729. In about half the cases, this results in no problems, with
>>> audio being encoded with G711. The other half of the time, they
>>> send us a second invite requesting G729. However, they proceed to
>>> send us a G711 encoded audio stream...
>>> 
>>> They have somewhat acknowledged the problem, but their advice is
>>> for us to only accept a single codec in our 200 OK. We don't want
>>> to disable either; we have customers using G729, so we'd like to
>>> avoid transcoding when possible, but we also do some T38 faxing,
>>> which I believe requires G711 to start off.
>>> 
>>> My first thought was to selectively force the codec on inbound
>>> calls - if it is for a voice number, use 729, otherwise 711.
>>> However, I can't find any way of doing this within Asterisk. (We do
>>> have an OpenSIPS server sitting between us and the provider, and I
>>> could use OpenSIPS features to do this; however, right now the
>>> OpenSIPS server is fairly dumb - it's only proxying traffic between
>>> us and the provider and knows nothing about our specific DIDs.)
>>> 
>>> A couple more details in case anyone has seen a similar issue. The
>>> provider is Broadvox, and this issue only seems to manifest on
>>> calls coming to them via Skype. They claim to not have any direct
>>> link with Skype, but it seems odd that the problem would be
>>> specific to Skype callers if the call is coming to Broadvox as a
>>> standard PSTN call.
>>> 
>>> Is there any way to do this? Am I totally missing something and
>>> making a stupid mistake, or making the issue more complicated than
>>> it needs to be?
>>> 
>> The problem here is that you have a proxy in between, so Asterisk
>> can't have separate peer configurations, since all the SIP messages
>> are from the same IP and thus the same peer. I have a branch that
>> implements peer matching in this specific configuration, which means
>> that you can have different codec configurations for different
>> partners even though there's a proxy in front of Asterisk.
>> 
>> https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4
>> 
>> Please try this branch and give feedback. There should be some docs
>> in sip.conf for the new "matchrule" setting.
>> 
>> /O
> 
> 
> I'd be interested in trying this out - but the site doesn't seem to be
> responding. :)
Sorry, gave you the developer URL. Too quick copy and paste...
Here's a correct one:
>> http://svn.digium.com/svn/asterisk/team/oej/pinetree-1.4


/O

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

[Olle E. Johansson]

21 mar 2010 kl. 18.22 skrev Philipp von Klitzing:

> Hi Olle!
> 
>> The work I started during Christmas - Named ACL's - is a starting point
>> that other developers can use to develop all kind of schemes.
>> 
>> http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists
>> -asterisk-nacls/
> 
> Very interesting. Doesn't look like this has any chance to secure 1.4 
> installations though, I am afraid.

The code was written both for trunk and 1.4. It won't be included in 1.4 
release though, right.

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

[Olle E. Johansson]

19 mar 2010 kl. 03.41 skrev Philipp von Klitzing:

> Hey hey!
> 
>>> My first step will be to strengthen the passwords in use, and for the
>>> hardphones to restrict by IP address, but that still leaves the
>>> softphone quite widely open.
>> 
>> Asterisk doesn't differentiate between a hard phone and a soft phone.
> 
> Although: One could think about enhancing Asterisk security by allowing 
> only a (number of) specific SIP user agent header (vendor, model) for a 
> SIP account - next to a strong password, of course. Or implement 
> something more dynamic like: Read and lock the current (or first) user 
> agent string, and then ping the admin if that changes and request an un-
> lock/re-auth.
Those are interesting ideas. We could implement a timeout for registrations,
so that we only accept re-registrations while we have an active registration,
and if that expires only accept new registrations after a timeout.
This will delay access at reboots of the Asterisk server though.
> 
>>> Does Asterisk 1.6 have anything in it that can automatically block out
>>> an attacking IP, say if it receives several 20 or so failed attempts
>>> from that IP in x minutes?
> 
> It would still be important to have a sip.conf paramter in 1.4 that is 
> similar to "delayreject" in iax.conf! One of my system has been scanned 
> 3 times in the past days, and it takes just a little over a minute for a 
> 10.000 account registration scan.

The work I started during Christmas - Named ACL's - is a starting point
that other developers can use to develop all kind of schemes.

http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists-asterisk-nacls/

/O
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Re: [asterisk-users] SIP codec negotiation / manipulation

[Olle E. Johansson]

17 mar 2010 kl. 16.37 skrev Kevin Sandy:

> We're having an odd issue with codec negotiation from one of our SIP 
> providers. Here's the basic situation.
> 
> We receive an invite from them advertising support for G711, G729, and G723. 
> In our response, we send back that we support G711 and G729. In about half 
> the cases, this results in no problems, with audio being encoded with G711. 
> The other half of the time, they send us a second invite requesting G729. 
> However, they proceed to send us a G711 encoded audio stream...
> 
> They have somewhat acknowledged the problem, but their advice is for us to 
> only accept a single codec in our 200 OK. We don't want to disable either; we 
> have customers using G729, so we'd like to avoid transcoding when possible, 
> but we also do some T38 faxing, which I believe requires G711 to start off.
> 
> My first thought was to selectively force the codec on inbound calls - if it 
> is for a voice number, use 729, otherwise 711. However, I can't find any way 
> of doing this within Asterisk. (We do have an OpenSIPS server sitting between 
> us and the provider, and I could use OpenSIPS features to do this; however, 
> right now the OpenSIPS server is fairly dumb - it's only proxying traffic 
> between us and the provider and knows nothing about our specific DIDs.)
> 
> A couple more details in case anyone has seen a similar issue. The provider 
> is Broadvox, and this issue only seems to manifest on calls coming to them 
> via Skype. They claim to not have any direct link with Skype, but it seems 
> odd that the problem would be specific to Skype callers if the call is coming 
> to Broadvox as a standard PSTN call.
> 
> Is there any way to do this? Am I totally missing something and making a 
> stupid mistake, or making the issue more complicated than it needs to be?
> 
The problem here is that you have a proxy in between, so Asterisk can't have 
separate peer configurations, since all the SIP messages are from the same IP 
and thus the same peer. I have a branch that implements peer matching in this 
specific configuration, which means that you can have different codec 
configurations for different partners even though there's a proxy in front of 
Asterisk. 

https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4

Please try this branch and give feedback. There should be some docs in sip.conf 
for the new "matchrule" setting.

/O
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Re: [asterisk-users] SIP Trunk with "multiple" remote ip-addresses

[Olle E. Johansson]

12 mar 2010 kl. 12.01 skrev Klaus Darilion:

> 
> 
> Am 02.03.2010 13:29, schrieb Magnus Benngård:
>> Hi!
>> 
>> Did a setup of 2 peers as Klaus suggested, it worked thx!
>> 
>> Has anyone thought about the possibility to add multiple ip/hosts to
>> "host="?
>> 
>> I my case: "host=130.244.190.42,130.244.190.46" or
>> "host=sip-corporate1.tele2.se,sip-corporate2.tele2.se"
>> 
>> Step 1 could be to send to the first ip/host and accept from both.
>> 
>> Step 2 could be "round-robin" send if both are up and alive...
> 
> IMO this would be a nice feature.
Check my "peerfailover" branch.

> 
>> Btw, did try trunk version, no support for multiple SRV records there.
> 
> IIRC correctly there is a patch on the bugtracker for SRV handling, but 
> I do not know if that patch would fix this too.
I haven't seen that. Interesting.

/O
> 
> regards
> klaus
> 
>> 
>> 
>> 
>>Am 02.03.2010 08:50, schrieb Magnus Benngård:
>>> Hi,
>>> 
>>> Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
>>> problem to get outgoing calls to work but i have some problems with
>>> incoming.
>>> 
>>> Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to
>>> "sip-corporate.tele2.se" which is either sip-corporate1.tele2.se
>>> (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46).
>>> 
>>> If i do a "sip show peer Tele2", I see that Asterisk has chosen
>>one of
>>> them: ToHost : sip-corporate.tele2.se
>>> Addr->IP : 130.244.190.46 Port 5060
>>> 
>>> Now my problems starts, when Tele2 sends a call to my Asterisk,
>>the call
>>> can come frome any of those two ip-adresses. If it comes from
>>> 130.244.190.46 everything if fine, but if it comes from
>>130.244.190.42:
>>> "[Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167
>>handle_request_invite:
>>> Failed to authenticate!"
>>> 
>>> I thought "srvlookup=yes" should take care about that, but then i
>>read a
>>> little bit more and found: "Note: Asterisk only uses the first
>>host in
>>> SRV records". :(
>> 
>>Hi Magnus!
>> 
>>Asterisk does not support multiple SRV records (expcet there were some
>>recent changes which I missed) - it takes one of the most priors and
>>use
>>it all the time.
>> 
>>Thus, in your scenario you have to specify the possible inbound sources
>>manually as peers:
>> 
>>[tele2-1]
>>type=peer
>>host=130.244.190.42
>>context=fromTele2
>>...
>>[tele2-2]
>>type=peer
>>host=130.244.190.46
>>context=fromTele2
>>...
>> 
>> 
>>regards
>>klaus
>> 
>> 
>>> 
>>> Can anyone plz give me some hint howto solve my problem?
>>> 
>>> Regards,
>>> 
>>> Magnus
>>> 
>> 
> 
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* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




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Re: [asterisk-users] dtmf payload 100

[Olle E. Johansson]

12 mar 2010 kl. 10.45 skrev Katerina Borin:

> Probably has anyone idea how dtmf payload type could be changed in Asterisk 
> say to 100? 
> 
> On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin  
> wrote:
> Hello,
> I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers 
> gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked 
> till  supplier has changed something. Now I receive from him dtmf payload 
> 100. With the second supplier which sends dtmf with payload type 101 
> everything works.
> 
> in cli I get this message as dtmf is entered
> rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP'
> 
> Is there any way to get asterisk understand dtmf payload type 100?
If they have declared it correctly in the SDP, we will understand. Since 
Asterisk doesn't recognize the codec, I belive they have a bug in their system.
In order for us to find out if Asterisk is doing wrong or if we can blame their 
system, we need to see the INVITE or 200 OK from their end. The information
you have provided here is not enough.

THanks,
/O
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Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

[Olle E. Johansson]

11 mar 2010 kl. 15.17 skrev Philipp von Klitzing:

>> Is there a way for a client to tell a server where it is registered to
>> remove the registration?
> 
> Yes, it needs to send an UNREGISTER sip message.
> 
There's actually not an UNREGISTER method in SIP.
As Kevin stated, you send a REGISTER with a zero expiry to cancel a current 
registration.

/O
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Re: [asterisk-users] Asterisk Management API

[Olle E. Johansson]

8 mar 2010 kl. 11.13 skrev Peter Childs:

> On 5 March 2010 13:48, Jim Dickenson  wrote:
>> At an Asterisk CLI use the command "manager show commands".
> 
> 
> Life is rarely that simple, and this does not really answer the question.
> 
> Oh and Channel can mean different things in different contexts
> 
> ie
> 
> Channel in a PlayDTMF command means a "Call" to play the DTMF on,
> where as Channel in a Originate command means the "Device to place the
> call on" so you can't use the same input for both commands (or can
> you?)

I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but 
not all. And the changes hurted a lot of existing applications, so I'm careful 
not to mess around too much with AMI again. The most important part is that we 
don't allow reuse of existing headers for new things in new actions and events. 
I've been trying to watch over manager in order to disallow misuse, but 
development is fast and it's easy to miss a commit or a review...

/O
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[asterisk-users] SIPit 26 in Sweden - organized by Edvina

[Olle E. Johansson]

Friends,

SIPit is the main interoperability event for all things SIP. It's organized by 
the SIP Forum and creates good
feedback to the IETF. Asterisk has been participating in SIPit during many 
years and in many variants 
- videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. 

All these tests has lead to a large amount of improvements for Asterisk and 
have helped us to build a
network with other developers in the business, a network which helps when we 
have bugs that involve
interoperability with these devices or servers. SIPit is important for 
Asterisk, and thus it is important for 
everyone in the Asterisk community. Now, when we are working on the next  
long-term release (1.8) we 
really need to test and make sure that we interoperate. New stuff, like Terry's 
SRTP branch and the
call completion and caller ID update work needs testing.

* Interoperability drives the TCP/IP business

What drives the TCP/IP business is simply stated in one word - 
"INTEROPERABILITY". Without open
network standards, the VoIP business would not be as large as it is today. 
Without working and tested
standards, it would not work at all. Asterisk, as an Open Source platform is in 
the middle of this business. 
We simply have to interoperate with all kinds of phones, servers and services 
out there. 

* SIPit 26 - organized by Edvina

SIPit is organized by the SIP Forum and every SIPit - two per year - is hosted 
by a company. During
good times, the large vendors has taken care of this. In the current climate, 
it's hard to get the needed
resources - time and money - from these vendors. SIPit 25, yes the 25th in a 
successful series, was
organized by the Interoperability Labs at University of New Hampshire in 
September 2009. Digium helped
out by providing the PBX infrastructure for the event. The next one, SIPit 26, 
will be in Stockholm, Sweden 
- hosted by myself with support from Tandberg and sponsored by a set of 
companies.

* SIP interoperability is a requirement - for you, your customers, your business

So why I am doing this? The business needs it, Asterisk needs it and I need it. 
Without SIPit, you will not
get good products that work together. Without SIPit, tests will be limited to 
certification of a very limited set
of functions by different vendors. I believe that this will lead to unhealthy 
market domination in the 
implementations, something that does not benefit the customers. 
And besides, after creating Astricon in the US and a series of other 
conferences in Sweden, I have 
experience of organizing events. I am crazy enough to step forward and take the 
risk, since I really
strongly believe that we all benefit from this.

Now I need your support too.

* Additions to your already filled-up TODO-list:

- If you develop SIP software, make sure you register and attend.
- If you buy SIP devices or software, ask the vendors if they attend SIPit and 
encourage them to participate
- If you have collegues that work with SIP development, please forward this 
mail to them!

Short facts:
- Date: May 17-21 2010 (very beautiful season in Sweden!)
- Location: Kista, Stockholm, Sweden
- Host: Edvina and Tandberg
- Sponsors: Ingate, Intertex, .se, Telio, Snom

We are currently working to set the price and to be able open for registration. 
SIPit 26 has a Facebook 
event page at http://www.facebook.com/event.php?eid=340634688354 and a Twitter 
stream: http://twitter.com/sipit26 where you will get updates and be able to 
find links to the host web 
site when it opens. The main web site for  SIPit is http://www.sipit.net - a 
site that explains what will 
happen during this week and why you should attend as a developer.

If you have any questions or suggestions, please don't hesitate to contact me.

Thanks for your support!

Regards,
/Olle




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Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Olle E. Johansson]

27 feb 2010 kl. 08.26 skrev Olle E. Johansson:

> 
> 26 feb 2010 kl. 22.02 skrev JT:
> 
>> Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat 
>> of a band-aid to the issue.  But in my observations there is one clear 
>> indicator that I am shocked is not used.
>> 
>> When I have done this test - pulling the network cable on a device during a 
>> call - Asterisk actually reports that the SIP device has become unreachable 
>> within seconds of the device's removal.
>> 
>> Now one would think, just like a regular phone company, if one device became 
>> unresponsive (unreachable), the call would be automatically dropped.  Like 
>> unplugging from a POTS while on a call.
>> 
>> So why would Asterisk not use the following logic:  
>> Is Device reachable?
>> Yes - Do nothing
>> 
>> No - Close all calls bridged to device
>> 
>> Seems that would solve the issue quickly and cleanly... perhaps with the RTP 
>> timeout being an additional measure of safety
>> 
>> Is this an issue present in the latest version of Asterisk?  My hope was it 
>> was simply an older bug, fixed at some later trunk.
> 
> 
> If there's a reason to send SIP messages during the call and they fail, the 
> call WILL be hung up.
> Reading the 1.4 RTP source code, I don't think we're checking the return 
> codes of the network writes.
> Now,  that can be very tricky. For a call with NAT, we will have to send 
> packets that fail until we receive something from the other end. I am just 
> brainstorming here, but we could have a flag set when we've received RTp 
> packets from the other end and from that moment start reacting on the result 
> codes of the sendto() call. If it's indicating network issues, we could 
> possibly have an option to tear the call down after a certain amount of 
> failures.
> 
> And no, I can't explain why someone hasn't thought of that. I think it would 
> be a good addition.

And after a few hours of hacking I know more. If the incoming channel dies, 
there will be no attempts at sending, so we won't have any network issues at 
all. The RTP channel in Asterisk is clocked on incoming media. The RTP timeouts 
we have today is the only solution for normally bridged calls.

The p2p rtp bridge behaves a bit differently and I think I found a bug in it, 
so I will have to investigate that part a bit more.

Now, we could hang up calls based on device status if needed. I have part of 
that code in the peerfailover branch.

/O
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Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Olle E. Johansson]

26 feb 2010 kl. 22.02 skrev JT:

> Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of 
> a band-aid to the issue.  But in my observations there is one clear indicator 
> that I am shocked is not used.
> 
> When I have done this test - pulling the network cable on a device during a 
> call - Asterisk actually reports that the SIP device has become unreachable 
> within seconds of the device's removal.
> 
> Now one would think, just like a regular phone company, if one device became 
> unresponsive (unreachable), the call would be automatically dropped.  Like 
> unplugging from a POTS while on a call.
> 
> So why would Asterisk not use the following logic:  
> Is Device reachable?
> Yes - Do nothing
> 
> No - Close all calls bridged to device
> 
> Seems that would solve the issue quickly and cleanly... perhaps with the RTP 
> timeout being an additional measure of safety
> 
> Is this an issue present in the latest version of Asterisk?  My hope was it 
> was simply an older bug, fixed at some later trunk.


If there's a reason to send SIP messages during the call and they fail, the 
call WILL be hung up.
Reading the 1.4 RTP source code, I don't think we're checking the return codes 
of the network writes.
Now,  that can be very tricky. For a call with NAT, we will have to send 
packets that fail until we receive something from the other end. I am just 
brainstorming here, but we could have a flag set when we've received RTp 
packets from the other end and from that moment start reacting on the result 
codes of the sendto() call. If it's indicating network issues, we could 
possibly have an option to tear the call down after a certain amount of 
failures.

And no, I can't explain why someone hasn't thought of that. I think it would be 
a good addition.

/O





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Re: [asterisk-users] directrtp with SIP + H.323

[Olle E. Johansson]

24 feb 2010 kl. 01.22 skrev Kristian Kielhofner:

> On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis  wrote:
>> We're creating a SIP gateway for a client that will take one leg of a call
>> in via SIP, and out the other side via H.323.  To minimize load on the
>> gateway, we would like to have the RTP stream bypass the gatewayy altogether
>> (directrtp/reinvite).  Is this possible with these to protocols?
>> 
>> Thanks
> 
> Yate claims it can do this:
> 
> http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
> 
There are two ways - either by reinvites, which according to Kevin won't work 
with H323, or by doing it right in the call setup. If we did that, we would 
stumble into the same problem as we have with this function in SIP - which goes 
all back to the media negotiation framework (see 
http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/
 ).

Asterisk currently just communicates an answered call as "answered" over the 
bridge without any attributes. This is the reason why the code has been marked 
"experimental" for many releases and no one has solved it. In order for this to 
work, you either need exactly the same codec attributes or a way to handle the 
ANSWER control frame (like John Martin did in the videocaps branch).

The hooks are all there if you want to experiment with this in the H.323 
channel. It's certainly possible. But it is not a function I would support 
generally (which is why the directrtp call setup function remains experimental).

/O
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