Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

Hmmm.. So if you are sure that the poke packets leave the network interface
(I would still check with tcpdump as well, maybe a firewall issue?) then it
makes sense to check the other side to make sure the poke packets reach
other servers.
I mean with tcpdump you may see if there are incoming packets from your
peers on the interface. If there are, then they are dropped or ignored by
your servers. If no, then it's better to check the other side.
you may try smth like 'tcpdump -npi  host  and port 4569'
Do you have a firewall configured on this server?

On Fri, Apr 21, 2017, 12:36 AM Carlos Chavez  wrote:

> On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:
>
> If SIP goes to the same provider then yes. Still I would check a packet
> capture for better understanding. BTW, did you try iax debug?
>
> чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :
>
>> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>>
>> Can it happen that the routes lead the traffic through another interface?
>> Did you try a packet capture with tcpdump? Do the packets really leave the
>> usb adapter? Can asymmetric routing be in effect?
>> Maybe there were some static routes that disappeared when the adapter was
>> unplugged...
>>
>> On Thu, Apr 20, 2017, 12:41 AM Antony Stone <
>> antony.st...@asterisk.open.source.it> wrote:
>>
>>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>>
>>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>>> > >
>>> > > You say the USB ethernet adapter got unplugged and then
>>> reconnected...
>>> > >
>>> > > 1. What's the name of the network device for this adapter?  Is it the
>>> > > same name as it previously had?
>>> > >
>>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>>> > >
>>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>>> > > 'bindport'?
>>> > >
>>> > > 4. Do you have SIP connections on the same network interface, and are
>>> > > those working as normal?
>>> > >
>>> > >
>>> > > Antony.
>>> >
>>> > 1- No changes to device names.  eth0 is the main link to the network,
>>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>>> > goes to another SIP provider.  All IAX trunks use eth0
>>> >
>>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>>> >
>>> > 3- We do not specify bindaddr or bindport in the config file as the
>>> > default is to bind to 0.0.0.0
>>> >
>>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>>> > servers.  The SIP trunk is working with no problems.  Except for two
>>> SIP
>>> > links to PSTN all internal extensions use the same network interface.
>>>
>>> Ugh :(
>>>
>>> Sorry, I have no more ideas, then.
>>>
>>> I hope someone else comes into this thread with a helpful suggestion.
>>>
>>>
>>> If routing was the problem then the SIP trunk would not work.
>> Usually IAX2 is a little more forgiving about routing than SIP.
>>
>> The new SIP trunks are replacing the IAX2 trunks to our other
> Asterisk servers and use exactly the same network paths, that is why I know
> it is not a network infrastructure issue.  We did turn on IAX debug and we
> only se the server trying to poke the other servers but there is not
> response or any incoming traffic.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
>
> --
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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:

If SIP goes to the same provider then yes. Still I would check a 
packet capture for better understanding. BTW, did you try iax debug?


чт, 20 апр. 2017 г. в 19:46, Carlos Chavez >:


On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:


Can it happen that the routes lead the traffic through another
interface? Did you try a packet capture with tcpdump? Do the
packets really leave the usb adapter? Can asymmetric routing be
in effect?
Maybe there were some static routes that disappeared when the
adapter was unplugged...


On Thu, Apr 20, 2017, 12:41 AM Antony Stone
mailto:antony.st...@asterisk.open.source.it>> wrote:

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this
adapter?  Is it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this
adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for
'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network
interface, and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to
the network,
> eth1 (also internal) goes to a SIP provider and eth2 (the
USB adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config
file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks
to all
> servers.  The SIP trunk is working with no problems. 
Except for two SIP

> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful
suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


The new SIP trunks are replacing the IAX2 trunks to our other 
Asterisk servers and use exactly the same network paths, that is why I 
know it is not a network infrastructure issue.  We did turn on IAX debug 
and we only se the server trying to poke the other servers but there is 
not response or any incoming traffic.


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Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

If SIP goes to the same provider then yes. Still I would check a packet
capture for better understanding. BTW, did you try iax debug?

чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :

> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>
> Can it happen that the routes lead the traffic through another interface?
> Did you try a packet capture with tcpdump? Do the packets really leave the
> usb adapter? Can asymmetric routing be in effect?
> Maybe there were some static routes that disappeared when the adapter was
> unplugged...
>
> On Thu, Apr 20, 2017, 12:41 AM Antony Stone <
> antony.st...@asterisk.open.source.it> wrote:
>
>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>
>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>> > >
>> > > You say the USB ethernet adapter got unplugged and then reconnected...
>> > >
>> > > 1. What's the name of the network device for this adapter?  Is it the
>> > > same name as it previously had?
>> > >
>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>> > >
>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>> > > 'bindport'?
>> > >
>> > > 4. Do you have SIP connections on the same network interface, and are
>> > > those working as normal?
>> > >
>> > >
>> > > Antony.
>> >
>> > 1- No changes to device names.  eth0 is the main link to the network,
>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>> > goes to another SIP provider.  All IAX trunks use eth0
>> >
>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>> >
>> > 3- We do not specify bindaddr or bindport in the config file as the
>> > default is to bind to 0.0.0.0
>> >
>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>> > servers.  The SIP trunk is working with no problems.  Except for two SIP
>> > links to PSTN all internal extensions use the same network interface.
>>
>> Ugh :(
>>
>> Sorry, I have no more ideas, then.
>>
>> I hope someone else comes into this thread with a helpful suggestion.
>>
>>
>> If routing was the problem then the SIP trunk would not work.
> Usually IAX2 is a little more forgiving about routing than SIP.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez+52 (55)8116-9161 <+52%2055%208116%209161>
>
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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:

Can it happen that the routes lead the traffic through another 
interface? Did you try a packet capture with tcpdump? Do the packets 
really leave the usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that disappeared when the adapter 
was unplugged...



On Thu, Apr 20, 2017, 12:41 AM Antony Stone 
> wrote:


On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this adapter?  Is
it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network interface,
and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to the
network,
> eth1 (also internal) goes to a SIP provider and eth2 (the USB
adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> servers.  The SIP trunk is working with no problems. Except for
two SIP
> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


--
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+52 (55)8116-9161

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Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

Can it happen that the routes lead the traffic through another interface?
Did you try a packet capture with tcpdump? Do the packets really leave the
usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that disappeared when the adapter was
unplugged...

On Thu, Apr 20, 2017, 12:41 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>
> > On 4/19/17 4:23 PM, Antony Stone wrote:
> > >
> > > You say the USB ethernet adapter got unplugged and then reconnected...
> > >
> > > 1. What's the name of the network device for this adapter?  Is it the
> > > same name as it previously had?
> > >
> > > 2. What does 'ifconfig' say the IP address is for this adapter?
> > >
> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > > 'bindport'?
> > >
> > > 4. Do you have SIP connections on the same network interface, and are
> > > those working as normal?
> > >
> > >
> > > Antony.
> >
> > 1- No changes to device names.  eth0 is the main link to the network,
> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
> > goes to another SIP provider.  All IAX trunks use eth0
> >
> > 2- ifconfig gives the proper IP and netmask for all interfaces
> >
> > 3- We do not specify bindaddr or bindport in the config file as the
> > default is to bind to 0.0.0.0
> >
> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> > servers.  The SIP trunk is working with no problems.  Except for two SIP
> > links to PSTN all internal extensions use the same network interface.
>
> Ugh :(
>
> Sorry, I have no more ideas, then.
>
> I hope someone else comes into this thread with a helpful suggestion.
>
>
> Antony.
>
> --
> The first fifty percent of an engineering project takes ninety percent of
> the
> time, and the remaining fifty percent takes another ninety percent of the
> time.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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>
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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then reconnected...
> > 
> > 1. What's the name of the network device for this adapter?  Is it the
> > same name as it previously had?
> > 
> > 2. What does 'ifconfig' say the IP address is for this adapter?
> > 
> > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > 'bindport'?
> > 
> > 4. Do you have SIP connections on the same network interface, and are
> > those working as normal?
> > 
> > 
> > Antony.
> 
> 1- No changes to device names.  eth0 is the main link to the network,
> eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
> goes to another SIP provider.  All IAX trunks use eth0
> 
> 2- ifconfig gives the proper IP and netmask for all interfaces
> 
> 3- We do not specify bindaddr or bindport in the config file as the
> default is to bind to 0.0.0.0
> 
> 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> servers.  The SIP trunk is working with no problems.  Except for two SIP
> links to PSTN all internal extensions use the same network interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful suggestion.


Antony.

-- 
The first fifty percent of an engineering project takes ninety percent of the 
time, and the remaining fifty percent takes another ninety percent of the time.

   Please reply to the list;
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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/19/17 4:23 PM, Antony Stone wrote:


On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote:


On 4/19/17 4:09 PM, Antony Stone wrote:

On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:

I have a server that had been operating for a few years now with

IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
now say UNREACHABLE.

...snip...


So far the only thing different is that the receive queue for port 4569
is not zero like all the other servers:

udp   128760  0 0.0.0.0:45690.0.0.0:*

Basically all packets for IAX2 are getting stuck in the queue.
Any

suggestions?

Have you tried rebooting the router which connects this machine to the
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

  We have already tried that.  One of the servers that has an IAX
trunk to this server is on the same local network so that eliminates any
firewall/router in the way.  We disabled iptables just in case too.

Hm :(

You say the USB ethernet adapter got unplugged and then reconnected...

1. What's the name of the network device for this adapter?  Is it the same
name as it previously had?

2. What does 'ifconfig' say the IP address is for this adapter?

3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and 'bindport'?

4. Do you have SIP connections on the same network interface, and are those
working as normal?


Antony.

1- No changes to device names.  eth0 is the main link to the network, 
eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter) 
goes to another SIP provider.  All IAX trunks use eth0


2- ifconfig gives the proper IP and netmask for all interfaces

3- We do not specify bindaddr or bindport in the config file as the 
default is to bind to 0.0.0.0


4- We had to make new SIP trunks to replace the IAX2 trunks to all 
servers.  The SIP trunk is working with no problems.  Except for two SIP 
links to PSTN all internal extensions use the same network interface.


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote:

> On 4/19/17 4:09 PM, Antony Stone wrote:
> > On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:
> >>I have a server that had been operating for a few years now with
> >> 
> >> IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
> >> now say UNREACHABLE.
> > 
> > ...snip...
> > 
> >> So far the only thing different is that the receive queue for port 4569
> >> is not zero like all the other servers:
> >> 
> >> udp   128760  0 0.0.0.0:45690.0.0.0:*
> >> 
> >>Basically all packets for IAX2 are getting stuck in the queue.
> >>Any
> >> 
> >> suggestions?
> > 
> > Have you tried rebooting the router which connects this machine to the
> > Internet?
> > 
> > It sounds like a stale connection-tracking table entry to me.
> > 
> > 
> > Antony.
> 
>  We have already tried that.  One of the servers that has an IAX
> trunk to this server is on the same local network so that eliminates any
> firewall/router in the way.  We disabled iptables just in case too.

Hm :(

You say the USB ethernet adapter got unplugged and then reconnected...

1. What's the name of the network device for this adapter?  Is it the same 
name as it previously had?

2. What does 'ifconfig' say the IP address is for this adapter?

3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and 'bindport'?

4. Do you have SIP connections on the same network interface, and are those 
working as normal?


Antony.

-- 
"It would appear we have reached the limits of what it is possible to achieve 
with computer technology, although one should be careful with such statements; 
they tend to sound pretty silly in five years."

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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/19/17 4:09 PM, Antony Stone wrote:


On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:


   I have a server that had been operating for a few years now with
IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
now say UNREACHABLE.

...snip...


So far the only thing different is that the receive queue for port 4569 is
not zero like all the other servers:

udp   128760  0 0.0.0.0:45690.0.0.0:*

   Basically all packets for IAX2 are getting stuck in the queue. Any
suggestions?

Have you tried rebooting the router which connects this machine to the
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

We have already tried that.  One of the servers that has an IAX 
trunk to this server is on the same local network so that eliminates any 
firewall/router in the way.  We disabled iptables just in case too.


--
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+52 (55)8116-9161


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Re: [asterisk-users] IAX2 getting stuck

[Antony Stone]

On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:

>   I have a server that had been operating for a few years now with
> IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
> now say UNREACHABLE.

...snip...

> So far the only thing different is that the receive queue for port 4569 is
> not zero like all the other servers:
> 
> udp   128760  0 0.0.0.0:45690.0.0.0:*
> 
>   Basically all packets for IAX2 are getting stuck in the queue. Any
> suggestions?

Have you tried rebooting the router which connects this machine to the 
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

-- 
"Linux is going to be part of the future. It's going to be like Unix was."

 - Peter Moore, Asia-Pacific general manager, Microsoft

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Re: [asterisk-users] IAX2 problem for WAN connections

[jg]

I found a way that works. Essentially, I deleted the register lines and added the hosts with 
deny all and specific permit specs. I don't know why it works, but it does.


jg

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Re: [asterisk-users] IAX2 problem for WAN connections

[jg]

On Thursday 05 Feb 2015, jg wrote:

Calling from ServerB to ServerA works, but not vice versa. The only odd
thing that appears to me is the different perceived port on ServerA.

Does someone have an idea at what to look in detail?


Look in /etc/asterisk/iax.conf in the first instance.



Basically I used the example from the Asterisk book "Connecting Two Asterisk Boxes Together via 
IAX" and there is not a lot to see:



; Server A
[general]
;   this boxremote IP
register => ServerA:very_sec...@80.152.xxx.xxx

disallow=all
allow=alaw
allow=ulaw
allow=gsm

jitterbuffer=no
forcejitterbuffer=no
autokill=yes


; the other box
[ServerB]
type=friend
trunk=no
auth=md5
encryption=yes
secret=very_secret
context=from-ServerB
qualify=yes
host=dynamic
; end of Server A


; Server B
[general]
;   this boxremote IP
register => ServerB:very_sec...@79.233.yyy.yyy

disallow=all
allow=alaw
allow=ulaw
allow=gsm

jitterbuffer=no
forcejitterbuffer=no
autokill=yes

; the other box
[ServerA]
type=friend
trunk=no
auth=md5
encryption=yes
secret=very_secret
context=from-ServerA
qualify=yes
host=dynamic
; end of Server B

If I replace the WAN addresses of the two routers with addresses on the LAN, everything works. 
Currently, I am not sure whether it could be a NAT related or Asterisk configuration problem.


jg

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Re: [asterisk-users] IAX2 problem for WAN connections

[A J Stiles]

On Thursday 05 Feb 2015, jg wrote:
> Calling from ServerB to ServerA works, but not vice versa. The only odd
> thing that appears to me is the different perceived port on ServerA.
> 
> ServerA*CLI> iax2 show registry
> Host  dnsmgr  Username PerceivedRefresh  State
> 80.152.xxx.xxx:4569   N   ServerA 79.233.yyy.yyy:45697  60 
> Registered
> 
> ServerB*CLI> iax2 show registry
> Host  dnsmgr  Username Perceived   Refresh  State
> 79.233.yyy.yyy:4569   N   ServerB 79.233.yyy.yyy:4569  60  Request
> Sent
> 
> Does someone have an idea at what to look in detail?


Look in /etc/asterisk/iax.conf in the first instance.

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Re: [asterisk-users] IAX2 trunk with on demand Internet link

[A J Stiles]

On Monday 02 Feb 2015, spartan1...@hushmail.com wrote:
> Hi, I'm connecting 2 Asterisk servers with an IAX2 trunk. Trunk works
> fine in testing, no problems there but the Internet at server-A is an
> "on-demand" system that is based on the amount of http/https traffic
> going through it (or if the link is brought up manually/via scripting
> interface). As such there will be times that the link is
> downworkflow-wise this is not an issue (trunk for a specific
> purpose) but are there any issues with how IAX2 will behave?
> Specifically, when the link comes up will the IAX2 trunk reconnect
> automatically? If so, how long will it take for the trunk to reconnect
> (high-speed, low-latency link)? If it won't come up reliably, does
> anyone have suggestions on methods to force an IAX2 reload, etc.?

It should all just work fine.  We have two machines linked by an IAX2 trunk, 
one of which is a bit temperamental and keeps needing rebooting -- but it 
works fine, just coming up as and when it needs to.

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Re: [asterisk-users] IAX2 trunk on IPV6

[Matthew Jordan]

On Tue, Apr 29, 2014 at 1:06 AM, Xengis Khan  wrote:

> Hi,
> I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an
> ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only
> ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the
> client asterisk with the server asterisk as IAX2 peer and want to connect
> to the IPV6 ip. I bind the server with ipv6 and also sending the
> registration request from the client(peer) to the ipv6 address. But its not
> peering. following is the client's iax.conf
>
> register => peer1:peer1pass@[IPV6]:port
>
> [peer1]
> type=peer
> context=topeer
> username=peer1
> secret=peer1pass
> trunk=yes
> host=XXX.XXX.XXX.XXX
> port=
> disallow=all
> allow=g729:40,g723:30
> qualify=yes
>
> Also my confusion is what value will be in 'host' property. I assigned as
> host=[IPV6]...but it shows error.
> Can anyone help with this issue.
>
>
IAX2 does not support IPv6 in that version of Asterisk. IPv6 support was
added to chan_iax2 in Asterisk 12 [1].

[1] https://wiki.asterisk.org/wiki/display/AST/New+in+12

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] IAX2 Trunk Encryption

[Elliott W]

I have some additional information, using wireshark and the IAX2 RFC I
walked through the handshaking of the call.  I did three different calls,
RSA Authentication WITHOUT forcing encryption, RSA Authentication WITH
forcing encryption, and "Clear Text" Authentication WITH forcing
encryption.  Keeping in mind that ALL setups have a secret set and it
complains if you don't when forcing encryption with RSA authentication.

The RFC speaks to when the encryption information element is included in
the NEW packed that following the AUTHREQ packet ALL communications are to
be encrypted.  When I used JUST a secret without RSA authentication I was
able to see this behavior.

In ALL cases whether using RSA or clear text secrets the Encryption format
(0x2b) in the NEW packet was 0x8001, the RFC doesn't tell me what that one
is.  It only lists 0x0001 as AES-128.  But it is consistent across all the
different calls.
Now there are differences at the AUTHREQ packet.  The Authorization method
(0x0e) is 0x0004 RSA for the RSA ones and 0x0003 which is not listed in the
spec but is likely clear text for the one not using RSA.  This is clearly
telling me that they are authorizing differently.

As I said, with clear text authentication all packets following the AUTHREQ
are encrypted, with RSA authentication they are NOT, I see the AUTHREP
packet, the ACK of that packet and then depending on whether I had
specified that encryption was required or not an ACCEPT (not required) or
REJECT (was required).  And I shouldn't see anything after AUTHREQ.

When it is accepted the call completes correctly, albeit without the
encryption I desire.  When it is rejected it gives me Cause (0x16) of "No
authority found" and a Hangup Cause (0x2a) of "Facility not subscribed
(0x32)

*I am starting to think that this is a defect with the IAX2 protocol
implementation, I have seen NOTHING indicating that you cannot use RSA
authentication with IAX2 encryption BUT that does seem to be what IS
happening.  Nor have I found anything indicating this was a conscious
design decision.*

So does anyone have anything to add?  How would I get this addressed as a
bug?



On Mon, Apr 7, 2014 at 11:54 AM, Elliott W wrote:

> Any ideas?  Still hoping..
>
>
> On Sun, Apr 6, 2014 at 12:03 AM, Elliott W wrote:
>
>> I have.
>>
>> On the receiving side I had gotten:
>> [2014-04-05 23:28:12] WARNING[1832] chan_iax2.c: Rejected connect
>> attempt. No secret present while force encrypt enabled.
>>
>> I had no secret because I was using RSA authentication and didn't think I
>> needed it, so I added EXACTLY the same line on both sides (copy/paste).
>> Now I get:
>> [2014-04-05 23:30:42] NOTICE[1832] chan_iax2.c: Call Terminated, Incoming
>> call is unencrypted while force encrypt is enabled.
>>
>> On the sending side I really get nothing useful:
>> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] pbx.c: -- Executing
>> [s@macro-dialout-trunk:22] Dial("SIP/comp-in-ch01-0001", "
>> IAX2/ch01_ch02/1234,300,Ttr") in new stack
>> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] app_dial.c: -- Called
>> IAX2/ch01_ch02/1234
>> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] chan_iax2.c: -- Hungup
>> 'IAX2/ch01_ch02-17634'
>> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] app_dial.c: == Everyone
>> is busy/congested at this time (1:0/0/1)
>> I modified the extension and the trunk name for security reasons, but
>> without force encryption calls flow back and forth easily.
>>
>> These three directives exist on both sides:
>> encryption=yes
>> forceencryption=yes
>> secret=mysecretcode
>>
>> So I'm kind of at a loss, I can see the options set, I can see:
>> [2014-04-05 23:59:32] VERBOSE[1832] chan_iax2.c: -- Accepting
>> AUTHENTICATED call from xxx.yyy.zzz.aaa:
>> when I DON'T have the force encryption set, so I can't see what else I
>> need to do..
>>
>> CEW
>>
>>
>>
>>
>> On Fri, Apr 4, 2014 at 7:07 PM, Steve Totaro <
>> stot...@totarotechnologies.com> wrote:
>>
>>> Have you enabled IAX2 debugging and tried some test calls?
>>>
>>> Thanks,
>>> Steve T
>>>
>>>
>>>
>>> On Fri, Apr 4, 2014 at 6:59 PM, Elliott W 
>>> wrote:
>>>
 That answered my question as to whether it WAS encrypted, I think, and
 the answer is no, the credentials are but all the rest is not.  That just
 leaves the question of what I need to do to get it encrypted..

 Thanks.


 On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
 stot...@totarotechnologies.com> wrote:

> Wireshark.
>
>
>
> On Fri, Apr 4, 2014 at 11:13 AM, Elliott W <
> dig...@private-address.info> wrote:
>
>> Ok, I think I am 90%+ there.
>>
>> Note: the configuration or status is the same on both sides unless
>> otherwise noted.
>>
>> I am using RSA keys for authentication and the calls are coming
>> through as authenticated so I'm sure that part works.
>>
>> The peer shows the "(E)" next to the status in Asterisk Info for the
>> IAX2 peers
>>

Re: [asterisk-users] IAX2 Trunk Encryption

[Elliott W]

Any ideas?  Still hoping..


On Sun, Apr 6, 2014 at 12:03 AM, Elliott W wrote:

> I have.
>
> On the receiving side I had gotten:
> [2014-04-05 23:28:12] WARNING[1832] chan_iax2.c: Rejected connect attempt.
> No secret present while force encrypt enabled.
>
> I had no secret because I was using RSA authentication and didn't think I
> needed it, so I added EXACTLY the same line on both sides (copy/paste).
> Now I get:
> [2014-04-05 23:30:42] NOTICE[1832] chan_iax2.c: Call Terminated, Incoming
> call is unencrypted while force encrypt is enabled.
>
> On the sending side I really get nothing useful:
> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] pbx.c: -- Executing
> [s@macro-dialout-trunk:22] Dial("SIP/comp-in-ch01-0001", "
> IAX2/ch01_ch02/1234,300,Ttr") in new stack
> [2014-04-05 23:30:42] VERBOSE[2795][C-0002] app_dial.c: -- Called
> IAX2/ch01_ch02/1234
> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] chan_iax2.c: -- Hungup
> 'IAX2/ch01_ch02-17634'
> [2014-04-05 23:30:43] VERBOSE[2795][C-0002] app_dial.c: == Everyone is
> busy/congested at this time (1:0/0/1)
> I modified the extension and the trunk name for security reasons, but
> without force encryption calls flow back and forth easily.
>
> These three directives exist on both sides:
> encryption=yes
> forceencryption=yes
> secret=mysecretcode
>
> So I'm kind of at a loss, I can see the options set, I can see:
> [2014-04-05 23:59:32] VERBOSE[1832] chan_iax2.c: -- Accepting
> AUTHENTICATED call from xxx.yyy.zzz.aaa:
> when I DON'T have the force encryption set, so I can't see what else I
> need to do..
>
> CEW
>
>
>
>
> On Fri, Apr 4, 2014 at 7:07 PM, Steve Totaro <
> stot...@totarotechnologies.com> wrote:
>
>> Have you enabled IAX2 debugging and tried some test calls?
>>
>> Thanks,
>> Steve T
>>
>>
>>
>> On Fri, Apr 4, 2014 at 6:59 PM, Elliott W wrote:
>>
>>> That answered my question as to whether it WAS encrypted, I think, and
>>> the answer is no, the credentials are but all the rest is not.  That just
>>> leaves the question of what I need to do to get it encrypted..
>>>
>>> Thanks.
>>>
>>>
>>> On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
>>> stot...@totarotechnologies.com> wrote:
>>>
 Wireshark.



 On Fri, Apr 4, 2014 at 11:13 AM, Elliott W >>> > wrote:

> Ok, I think I am 90%+ there.
>
> Note: the configuration or status is the same on both sides unless
> otherwise noted.
>
> I am using RSA keys for authentication and the calls are coming
> through as authenticated so I'm sure that part works.
>
> The peer shows the "(E)" next to the status in Asterisk Info for the
> IAX2 peers
>
> The trunk configuration contains:
> encryption=yes
>
> So here is my question, Calls stop flowing when I use the directive:
> forceencryption=yes
> At the trunk level or higher does not matter, same effect.
>
> So my question comes down to, are my calls getting encrypted and why
> does this directive cause them to fail, AND how can I tell.
>
> Thanks.
>
>


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>>>
>>>
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Re: [asterisk-users] IAX2 Trunk Encryption

[Elliott W]

I have.

On the receiving side I had gotten:
[2014-04-05 23:28:12] WARNING[1832] chan_iax2.c: Rejected connect attempt.
No secret present while force encrypt enabled.

I had no secret because I was using RSA authentication and didn't think I
needed it, so I added EXACTLY the same line on both sides (copy/paste).
Now I get:
[2014-04-05 23:30:42] NOTICE[1832] chan_iax2.c: Call Terminated, Incoming
call is unencrypted while force encrypt is enabled.

On the sending side I really get nothing useful:
[2014-04-05 23:30:42] VERBOSE[2795][C-0002] pbx.c: -- Executing
[s@macro-dialout-trunk:22] Dial("SIP/comp-in-ch01-0001", "
IAX2/ch01_ch02/1234,300,Ttr") in new stack
[2014-04-05 23:30:42] VERBOSE[2795][C-0002] app_dial.c: -- Called
IAX2/ch01_ch02/1234
[2014-04-05 23:30:43] VERBOSE[2795][C-0002] chan_iax2.c: -- Hungup
'IAX2/ch01_ch02-17634'
[2014-04-05 23:30:43] VERBOSE[2795][C-0002] app_dial.c: == Everyone is
busy/congested at this time (1:0/0/1)
I modified the extension and the trunk name for security reasons, but
without force encryption calls flow back and forth easily.

These three directives exist on both sides:
encryption=yes
forceencryption=yes
secret=mysecretcode

So I'm kind of at a loss, I can see the options set, I can see:
[2014-04-05 23:59:32] VERBOSE[1832] chan_iax2.c: -- Accepting AUTHENTICATED
call from xxx.yyy.zzz.aaa:
when I DON'T have the force encryption set, so I can't see what else I need
to do..

CEW




On Fri, Apr 4, 2014 at 7:07 PM, Steve Totaro  wrote:

> Have you enabled IAX2 debugging and tried some test calls?
>
> Thanks,
> Steve T
>
>
>
> On Fri, Apr 4, 2014 at 6:59 PM, Elliott W wrote:
>
>> That answered my question as to whether it WAS encrypted, I think, and
>> the answer is no, the credentials are but all the rest is not.  That just
>> leaves the question of what I need to do to get it encrypted..
>>
>> Thanks.
>>
>>
>> On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
>> stot...@totarotechnologies.com> wrote:
>>
>>> Wireshark.
>>>
>>>
>>>
>>> On Fri, Apr 4, 2014 at 11:13 AM, Elliott W 
>>> wrote:
>>>
 Ok, I think I am 90%+ there.

 Note: the configuration or status is the same on both sides unless
 otherwise noted.

 I am using RSA keys for authentication and the calls are coming through
 as authenticated so I'm sure that part works.

 The peer shows the "(E)" next to the status in Asterisk Info for the
 IAX2 peers

 The trunk configuration contains:
 encryption=yes

 So here is my question, Calls stop flowing when I use the directive:
 forceencryption=yes
 At the trunk level or higher does not matter, same effect.

 So my question comes down to, are my calls getting encrypted and why
 does this directive cause them to fail, AND how can I tell.

 Thanks.


>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>
>>
>>
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>
>
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Re: [asterisk-users] IAX2 Trunk Encryption

[Steve Totaro]

Have you enabled IAX2 debugging and tried some test calls?

Thanks,
Steve T


On Fri, Apr 4, 2014 at 6:59 PM, Elliott W wrote:

> That answered my question as to whether it WAS encrypted, I think, and the
> answer is no, the credentials are but all the rest is not.  That just
> leaves the question of what I need to do to get it encrypted..
>
> Thanks.
>
>
> On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
> stot...@totarotechnologies.com> wrote:
>
>> Wireshark.
>>
>>
>>
>> On Fri, Apr 4, 2014 at 11:13 AM, Elliott W 
>> wrote:
>>
>>> Ok, I think I am 90%+ there.
>>>
>>> Note: the configuration or status is the same on both sides unless
>>> otherwise noted.
>>>
>>> I am using RSA keys for authentication and the calls are coming through
>>> as authenticated so I'm sure that part works.
>>>
>>> The peer shows the "(E)" next to the status in Asterisk Info for the
>>> IAX2 peers
>>>
>>> The trunk configuration contains:
>>> encryption=yes
>>>
>>> So here is my question, Calls stop flowing when I use the directive:
>>> forceencryption=yes
>>> At the trunk level or higher does not matter, same effect.
>>>
>>> So my question comes down to, are my calls getting encrypted and why
>>> does this directive cause them to fail, AND how can I tell.
>>>
>>> Thanks.
>>>
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
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>>
>
>
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Re: [asterisk-users] IAX2 Trunk Encryption

[Elliott W]

That answered my question as to whether it WAS encrypted, I think, and the
answer is no, the credentials are but all the rest is not.  That just
leaves the question of what I need to do to get it encrypted..

Thanks.


On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

> Wireshark.
>
>
>
> On Fri, Apr 4, 2014 at 11:13 AM, Elliott W wrote:
>
>> Ok, I think I am 90%+ there.
>>
>> Note: the configuration or status is the same on both sides unless
>> otherwise noted.
>>
>> I am using RSA keys for authentication and the calls are coming through
>> as authenticated so I'm sure that part works.
>>
>> The peer shows the "(E)" next to the status in Asterisk Info for the IAX2
>> peers
>>
>> The trunk configuration contains:
>> encryption=yes
>>
>> So here is my question, Calls stop flowing when I use the directive:
>> forceencryption=yes
>> At the trunk level or higher does not matter, same effect.
>>
>> So my question comes down to, are my calls getting encrypted and why does
>> this directive cause them to fail, AND how can I tell.
>>
>> Thanks.
>>
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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Re: [asterisk-users] IAX2 Trunk Encryption

[Steve Totaro]

Wireshark.


On Fri, Apr 4, 2014 at 11:13 AM, Elliott W wrote:

> Ok, I think I am 90%+ there.
>
> Note: the configuration or status is the same on both sides unless
> otherwise noted.
>
> I am using RSA keys for authentication and the calls are coming through as
> authenticated so I'm sure that part works.
>
> The peer shows the "(E)" next to the status in Asterisk Info for the IAX2
> peers
>
> The trunk configuration contains:
> encryption=yes
>
> So here is my question, Calls stop flowing when I use the directive:
> forceencryption=yes
> At the trunk level or higher does not matter, same effect.
>
> So my question comes down to, are my calls getting encrypted and why does
> this directive cause them to fail, AND how can I tell.
>
> Thanks.
>
>
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Re: [asterisk-users] IAX2 bridge failing

[Michelle Dupuis]

No - but this is a new setup so I can't say it worked before...it just isn't 
working from the start.

I've found the call setup works and once bridged there is one way audio (to the 
ATA, none from the ATA).  And the the connection drops after 30 secs approx 
because something on the path (or endpoint) realizes something is wrong...


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davis 
[stda...@multiservice.com]
Sent: Sunday, December 15, 2013 12:41 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Did you change your network switch recently?  Some Digium IAX ATAs do not 
behave well with Cisco equipment.


On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis 
mailto:mdup...@ocg.ca>> wrote:
meant to say restart didn't help either..


From: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 On Behalf Of Michelle Dupuis [mdup...@ocg.ca<mailto:mdup...@ocg.ca>]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Ok just restart

-Original Message-
From: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" 
inbetween)same result

I agree it sounds like something either end is using the wrong IP/port address 
somewhere in the call (yet signalling works fine).

Anything else to suggest?  I was hoping for an externalip type setting but not 
in iax2 (at least not in 1.4.x.x) 
From: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 On Behalf Of Joshua Colp [jc...@digium.com<mailto:jc...@digium.com>]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
> Some more details...I noticed that the call is bridged, and audio goes
> one way. However, the dial command still times out after 35 seconds
> (approx), and exists non-zero.
> While the channels are up, I did an core show channel xxx and found
> Blocking in:
> ast_waitfor_nandfds
> Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc 
in question, reload, retry, and see if that changes the behavior. If it does 
then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:  
www.digium.com<http://www.digium.com>  & 
www.asterisk.org<http://www.asterisk.org>

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Re: [asterisk-users] IAX2 bridge failing

[Steven Davis]

Did you change your network switch recently?  Some Digium IAX ATAs do not
behave well with Cisco equipment.


On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis  wrote:

> meant to say restart didn't help either..
>
> 
> From: asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [
> mdup...@ocg.ca]
> Sent: Saturday, December 14, 2013 11:20 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> Ok just restart
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
> Sent: Friday, December 13, 2013 11:46 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload"
> inbetween)same result
>
> I agree it sounds like something either end is using the wrong IP/port
> address somewhere in the call (yet signalling works fine).
>
> Anything else to suggest?  I was hoping for an externalip type setting but
> not in iax2 (at least not in 1.4.x.x)
> 
> From: asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp [
> jc...@digium.com]
> Sent: Friday, December 13, 2013 11:44 AM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] IAX2 bridge failing
>
> Michelle Dupuis wrote:
> > Some more details...I noticed that the call is bridged, and audio goes
> > one way. However, the dial command still times out after 35 seconds
> > (approx), and exists non-zero.
> > While the channels are up, I did an core show channel xxx and found
> > Blocking in:
> > ast_waitfor_nandfds
> > Is this a bug? Or something I can fix through config?
>
> Hola,
>
> Set "transfer=no" under the entries in iax.conf for the
> peers/users/friends/etc in question, reload, retry, and see if that changes
> the behavior. If it does then something involved may not like
> IAX2 native transfers.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
> www.digium.com  & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
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-- 
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VoIP Engineer
Multi Service

+1-913-663-9748 o
+1-913-871-5155 m

stda...@multiservice.com

<http://www.multiservice.com/>

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Re: [asterisk-users] IAX2 bridge failing

[Michelle Dupuis]

meant to say restart didn't help either..


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis 
[mdup...@ocg.ca]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Ok just restart

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" 
inbetween)same result

I agree it sounds like something either end is using the wrong IP/port address 
somewhere in the call (yet signalling works fine).

Anything else to suggest?  I was hoping for an externalip type setting but not 
in iax2 (at least not in 1.4.x.x) 
From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp 
[jc...@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
> Some more details...I noticed that the call is bridged, and audio goes
> one way. However, the dial command still times out after 35 seconds
> (approx), and exists non-zero.
> While the channels are up, I did an core show channel xxx and found
> Blocking in:
> ast_waitfor_nandfds
> Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc 
in question, reload, retry, and see if that changes the behavior. If it does 
then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:  
www.digium.com  & www.asterisk.org

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Re: [asterisk-users] IAX2 bridge failing

[Michelle Dupuis]

Ok just restart

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" 
inbetween)same result

I agree it sounds like something either end is using the wrong IP/port address 
somewhere in the call (yet signalling works fine).

Anything else to suggest?  I was hoping for an externalip type setting but not 
in iax2 (at least not in 1.4.x.x) 
From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp 
[jc...@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
> Some more details...I noticed that the call is bridged, and audio goes 
> one way. However, the dial command still times out after 35 seconds 
> (approx), and exists non-zero.
> While the channels are up, I did an core show channel xxx and found 
> Blocking in:
> ast_waitfor_nandfds
> Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc 
in question, reload, retry, and see if that changes the behavior. If it does 
then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:  
www.digium.com  & www.asterisk.org

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Re: [asterisk-users] IAX2 bridge failing

[Michelle Dupuis]

Some more details...I noticed that the call is bridged, and audio goes one way. 
 However, the dial command still times out after 35 seconds (approx), and 
exists non-zero.

While the channels are up, I did an core show channel xxx and found Blocking in:
ast_waitfor_nandfds

Is this a bug?  Or something I can fix through config?


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis 
[mdup...@ocg.ca]
Sent: Thursday, December 12, 2013 5:08 PM
To: Asterisk Users List
Subject: [asterisk-users] IAX2 bridge failing

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system.  The Asterisk 
system has been stable for years, and has no trouble bridge SIP phone sets to 
IAX trunks.

When I initiate a call from the IAX ATA, something goes wrong.One rare 
occasion it works fine, but usually there is no audio passed.  I have a snippet 
of the console below.  Notice no bridging message...not sure if that's a clue?  
The dialplan seems to execute properly, and I can watch the destination system 
which answers the call and starts playing media (monkeys) which I don't hear.

Any ideas on what is going on?  Since this is IAX in and IAX out, NAT should 
not be an issue (even through there is NAT on both sides).  Since media moves 
on the same UDP port as call setup, also proves should not be a network problem 
(I think)

Can someone point me to a solution?

Thanks!


(IP's and ISP and phone number disguised)

- Executing [s@macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in 
new stack
-- Goto (macro-dialexternal,s,60)
-- Executing [s@macro-dialexternal:60] Dial("IAX2/S-14468", 
"IAX2/ISP123/1234567890|60|W") in new stack
-- Called ISP123/1234567890
-- Call accepted by 201.191.37.138 (format ulaw)
-- Format for call is ulaw
-- IAX2/ISP123-2261 answered IAX2/S-14468
-- Channel 'IAX2/S-14468' ready to transfer
-- Channel 'IAX2/ISP123-2261' ready to transfer
-- Hungup 'IAX2/ISP123-2261'
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Re: [asterisk-users] IAX2 bridge failing

[Michelle Dupuis]

I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" 
inbetween)same result

I agree it sounds like something either end is using the wrong IP/port address 
somewhere in the call (yet signalling works fine).

Anything else to suggest?  I was hoping for an externalip type setting but not 
in iax2 (at least not in 1.4.x.x)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp 
[jc...@digium.com]
Sent: Friday, December 13, 2013 11:44 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing

Michelle Dupuis wrote:
> Some more details...I noticed that the call is bridged, and audio goes
> one way. However, the dial command still times out after 35 seconds
> (approx), and exists non-zero.
> While the channels are up, I did an core show channel xxx and found
> Blocking in:
> ast_waitfor_nandfds
> Is this a bug? Or something I can fix through config?

Hola,

Set "transfer=no" under the entries in iax.conf for the
peers/users/friends/etc in question, reload, retry, and see if that
changes the behavior. If it does then something involved may not like
IAX2 native transfers.

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] IAX2 bridge failing

[Joshua Colp]

Michelle Dupuis wrote:

Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is this a bug? Or something I can fix through config?


Hola,

Set "transfer=no" under the entries in iax.conf for the 
peers/users/friends/etc in question, reload, retry, and see if that 
changes the behavior. If it does then something involved may not like 
IAX2 native transfers.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] iax2: no authentication, but still peer?

[Sean Darcy]

On 10/08/2013 03:29 PM, Adrian Serafini wrote:

The qualify is on for the peer.  It is failing to reply to the requested
SIP status.  Maybe it is on wifi, screen goes off, wifi follows, zoiper
iax stack doesn't re-reg with the asterisk.


[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 441
[Oct 8 18:15:58] NOTICE[519]: chan_iax2.c:8153 register_verify: Host
 failed MD5 authentication for 'n4'
(c374d0a70c72e6e9bd359aa6a0f1a6c2 != 2c76c104bbfc3d54f566490f40cd12bd)
[Oct 8 18:19:17] NOTICE[517]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1002 ms)!
[Oct 8 18:19:29] NOTICE[512]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 300
[Oct 8 18:26:02] NOTICE[519]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1017 ms)!
ip-172-31-29-115*CLI> iax2 show peers
Name/Username Host Mask Port Status Description
n4  (D) 255.255.255.255 4569 LAGGED (1017 ms)

is it still registered, or do we really have an authentication problem?

sean





Got it thanks.  So the issue is that zoiper should re-register when the 
wifi comes back on.


sean


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Re: [asterisk-users] iax2: no authentication, but still peer?

[Adrian Serafini]

The qualify is on for the peer.  It is failing to reply to the requested 
SIP status.  Maybe it is on wifi, screen goes off, wifi follows, zoiper 
iax stack doesn't re-reg with the asterisk.


[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 441
[Oct 8 18:15:58] NOTICE[519]: chan_iax2.c:8153 register_verify: Host
 failed MD5 authentication for 'n4'
(c374d0a70c72e6e9bd359aa6a0f1a6c2 != 2c76c104bbfc3d54f566490f40cd12bd)
[Oct 8 18:19:17] NOTICE[517]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1002 ms)!
[Oct 8 18:19:29] NOTICE[512]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 300
[Oct 8 18:26:02] NOTICE[519]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1017 ms)!
ip-172-31-29-115*CLI> iax2 show peers
Name/Username Host Mask Port Status Description
n4  (D) 255.255.255.255 4569 LAGGED (1017 ms)

is it still registered, or do we really have an authentication problem?

sean





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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/10/2013 12:15 PM, Joshua Colp wrote:

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.



Since no horse is dead enough not to take another beating:

If the console is showing the *source* port of the packets. then:

does the server send iax packets to that source port, or to 4569?

"home" (which is another asterisk server) shows 4569, while the androids 
running zoiper show random ports. I assume zoiper puts the source port 
in an iax packet. But regardless of how zoiper describes its source 
port, asterisk will only send iax packets on 4569. correct?


I ask all this because Amazon EC2 uses a firewall that doesn't have a 
connection state. All incoming ports are blocked unless they are 
explicitly opened. Just having a packet go out to an ip address and 
port, doesn't open the source port.


But if iax is always and only using 4569 to send and receive, I don't 
have to worry about opening any other ports.


Thanks,

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/10/2013 05:27 PM, Joshua Colp wrote:

Sean Darcy wrote:

On 09/10/2013 12:15 PM, Joshua Colp wrote:

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.



Since no horse is dead enough not to take another beating:

If the console is showing the *source* port of the packets. then:

does the server send iax packets to that source port, or to 4569?


It sends to the source port if using the registration.



"home" (which is another asterisk server) shows 4569, while the androids
running zoiper show random ports. I assume zoiper puts the source port
in an iax packet. But regardless of how zoiper describes its source
port, asterisk will only send iax packets on 4569. correct?


It does not put the source port in an IAX packet. It's in the IP header
itself, outside of IAX. Asterisk will send IAX packets *from* port 4569
but *to* any host/port.



OK, so I only need to open up 4569 incoming, But I need to allow a range 
of outgoing udp ports since zoiper is choosing other udp ports in the IP 
header of the iax registration.


Thanks. Sorry it's taken so long for me to get this.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Joshua Colp]

Sean Darcy wrote:

On 09/10/2013 12:15 PM, Joshua Colp wrote:

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.



Since no horse is dead enough not to take another beating:

If the console is showing the *source* port of the packets. then:

does the server send iax packets to that source port, or to 4569?


It sends to the source port if using the registration.



"home" (which is another asterisk server) shows 4569, while the androids
running zoiper show random ports. I assume zoiper puts the source port
in an iax packet. But regardless of how zoiper describes its source
port, asterisk will only send iax packets on 4569. correct?


It does not put the source port in an IAX packet. It's in the IP header 
itself, outside of IAX. Asterisk will send IAX packets *from* port 4569 
but *to* any host/port.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 07:48 PM, Eric Wieling wrote:

Try this as an example of why it doesn't matter.

1) On windows open a cmd prompt or on linux open up a local terminal.
2) open a web browser and connect to a web site like cnn.com
3) on windows type "netstat -n" in the command prompt, in linux type netstat -n 
--ip

For example on my system, the local IP is 172.17.3.111.  Notice below how the 
port on my local system is NOT 80, even though the port on the remote system 
is?   This is simply how TCP and UDP work.  When you are looking at your iax 
peers you are seeing the REMOTE IP and REMOTE port, which seldom matters.  It 
is the port on the client you are connecting TO which matters, not the port 
which you are connecting FROM. TCP and UDP do not allow more than one 
connection using the same source IP/source port/destination IP/destination port 
(called a tuple).  For most things the source port does not matter so the 
operating system assigns whatever source port it wants to.   NAT routers will 
often change the source port when the connection is NAT'd.  These are 
fundamental IP networking concepts whi
  ch all people doing VoIP should know, but most don't. I'm sure there are 
many books on TCP/IP networking which explain it better than I have explained 
it.

Active Connections

   Proto  Local Address  Foreign AddressState
TCP172.17.3.111:22020 157.166.226.25:80  ESTABLISHED
  TCP172.17.3.111:22021 157.166.249.10:80  ESTABLISHED
  TCP172.17.3.111:22022 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22023 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22024 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22025 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22026 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22027 23.203.4.211:80ESTABLISHED
  TCP172.17.3.111:22028 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22029 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22030 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22031 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22032 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22033 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22034 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22035 74.217.240.83:80   ESTABLISHED
  TCP172.17.3.111:22036 23.63.227.123:80   ESTABLISHED
  TCP172.17.3.111:22037 12.130.81.225:80   ESTABLISHED
  TCP172.17.3.111:22038 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22039 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22040 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22041 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22042 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22043 4.26.252.126:80ESTABLISHED

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 7:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

On 09/09/2013 03:37 PM, Eric Wieling wrote:

Again, that port is assigned by your NAT router.  Asterisk cannot control the 
source port if the incoming packet.   That is set by your NAT router and client 
and likely has nothing to do with your problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean
Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from
same ip address

Dial("IAX2/home-14358", "IAX2/gn") in new stack
   -- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port
Status  Description
gn (D)  255.255.255.255  9007  OK
(179 ms)

[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max retries 
exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 11, 
ts=10018, seqno=1)
   -- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought all iax 
traffic went over 4569.

Of course, this could be a zoiper problem.

sean



But the problem is it's not MY nat router; it's amazon's. And if you only have only have 
one iax device registered, it's always 4569, So why does amazon assign a different port 
to the second iax device? How would it even "know"?

sean



Well, I may be confused, but iax show peers is showing the remote port, 
the port it will connect TO, right?


netstat doesn't show the asterisk connections at all, ju

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Joshua Colp]

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default, 
binds to a single UDP port (4569). As UDP is connectionless there are no 
connections. What you see on the console is the *source* IP address and 
port of the packets. It's possible that the Amazon stuff is sort of 
NATting things to do connection tracking... but that's Amazon land, so 
no clue really.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Eric Wieling]

Try this as an example of why it doesn't matter.

1) On windows open a cmd prompt or on linux open up a local terminal.
2) open a web browser and connect to a web site like cnn.com
3) on windows type "netstat -n" in the command prompt, in linux type netstat -n 
--ip

For example on my system, the local IP is 172.17.3.111.  Notice below how the 
port on my local system is NOT 80, even though the port on the remote system 
is?   This is simply how TCP and UDP work.  When you are looking at your iax 
peers you are seeing the REMOTE IP and REMOTE port, which seldom matters.  It 
is the port on the client you are connecting TO which matters, not the port 
which you are connecting FROM. TCP and UDP do not allow more than one 
connection using the same source IP/source port/destination IP/destination port 
(called a tuple).  For most things the source port does not matter so the 
operating system assigns whatever source port it wants to.   NAT routers will 
often change the source port when the connection is NAT'd.  These are 
fundamental IP networking concepts which all people doing VoIP should know, but 
most don't. I'm sure there are many books on TCP/IP networking which 
explain it better than I have explained it.

Active Connections

  Proto  Local Address  Foreign AddressState
TCP172.17.3.111:22020 157.166.226.25:80  ESTABLISHED
 TCP172.17.3.111:22021 157.166.249.10:80  ESTABLISHED
 TCP172.17.3.111:22022 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22023 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22024 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22025 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22026 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22027 23.203.4.211:80ESTABLISHED
 TCP172.17.3.111:22028 23.63.227.185:80   ESTABLISHED
 TCP172.17.3.111:22029 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22030 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22031 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22032 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22033 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22034 4.27.18.126:80 ESTABLISHED
 TCP172.17.3.111:22035 74.217.240.83:80   ESTABLISHED
 TCP172.17.3.111:22036 23.63.227.123:80   ESTABLISHED
 TCP172.17.3.111:22037 12.130.81.225:80   ESTABLISHED
 TCP172.17.3.111:22038 4.26.252.126:80ESTABLISHED
 TCP172.17.3.111:22039 4.26.252.126:80ESTABLISHED
 TCP172.17.3.111:22040 4.26.252.126:80ESTABLISHED
 TCP172.17.3.111:22041 4.26.252.126:80ESTABLISHED
 TCP172.17.3.111:22042 4.26.252.126:80ESTABLISHED
 TCP172.17.3.111:22043 4.26.252.126:80ESTABLISHED

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 7:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

On 09/09/2013 03:37 PM, Eric Wieling wrote:
> Again, that port is assigned by your NAT router.  Asterisk cannot control the 
> source port if the incoming packet.   That is set by your NAT router and 
> client and likely has nothing to do with your problem.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean 
> Darcy
> Sent: Monday, September 09, 2013 3:30 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] iax2: two users can't authenticate from 
> same ip address
>
> Dial("IAX2/home-14358", "IAX2/gn") in new stack
>   -- Called IAX2/gn
> CLI> iax2 show peers
> Name/UsernameHost Mask Port
> Status  Description
> gn (D)  255.255.255.255  9007  OK
> (179 ms)
> 
> [Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max 
> retries exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 
> 11, ts=10018, seqno=1)
>   -- Hungup 'IAX2/gn-11311'
>
> Again, what's with this port 9007? Is asterisk assigning it? I thought all 
> iax traffic went over 4569.
>
> Of course, this could be a zoiper problem.
>
> sean
>

But the problem is it's not MY nat router; it's amazon's. And if you only have 
only have one iax device registered, it's always 4569, So why does amazon 
assign a different port to the second iax device? How would it even "know"?

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 03:37 PM, Eric Wieling wrote:

Again, that port is assigned by your NAT router.  Asterisk cannot control the 
source port if the incoming packet.   That is set by your NAT router and client 
and likely has nothing to do with your problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

Dial("IAX2/home-14358", "IAX2/gn") in new stack
  -- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port
Status  Description
gn (D)  255.255.255.255  9007  OK
(179 ms)

[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max retries 
exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 11, 
ts=10018, seqno=1)
  -- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought all iax 
traffic went over 4569.

Of course, this could be a zoiper problem.

sean



But the problem is it's not MY nat router; it's amazon's. And if you 
only have only have one iax device registered, it's always 4569, So why 
does amazon assign a different port to the second iax device? How would 
it even "know"?


sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Joshua Colp]

Sean Darcy wrote:


home is from the home machine, which registers with the server:

register => home:@

[home]
type=friend
insecure=port,invite
secret= ; same secret as on server
context=incoming
host=


You aren't specifying what username to authenticate as here. Add:

username=home

And give it a go.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Eric Wieling]

Again, that port is assigned by your NAT router.  Asterisk cannot control the 
source port if the incoming packet.   That is set by your NAT router and client 
and likely has nothing to do with your problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

Dial("IAX2/home-14358", "IAX2/gn") in new stack
 -- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port 
Status  Description
gn (D)  255.255.255.255  9007  OK 
(179 ms)

[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max 
retries exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 11, 
ts=10018, seqno=1)
 -- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought all iax 
traffic went over 4569.

Of course, this could be a zoiper problem.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 01:54 PM, Joshua Colp wrote:

Sean Darcy wrote:


home is from the home machine, which registers with the server:

register => home:@

[home]
type=friend
insecure=port,invite
secret= ; same secret as on server
context=incoming
host=


You aren't specifying what username to authenticate as here. Add:

username=home

And give it a go.



Excellent! It's so easy to overlook the obvious.

But now I can't call "gn". I can call out from gn, but calling to gn 
dies with Max retries...


Dial("IAX2/home-14358", "IAX2/gn") in new stack
-- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port 
Status  Description
gn (D)  255.255.255.255  9007  OK 
(179 ms)


[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max 
retries exceeded to host  on IAX2/gn-11311 (type = 6, subclass 
= 11, ts=10018, seqno=1)

-- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought 
all iax traffic went over 4569.


Of course, this could be a zoiper problem.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 11:08 AM, Joshua Colp wrote:

Sean Darcy wrote:


On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other
peers are assigned 1026, 1027 and 1028. How are these ports assigned?


The actual configuration entries (minus password) for each one involved
would be useful... if you aren't being explicit with what username to
use for outgoing authentication then stuff like this can happen.



On the server:

[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2

[gn](default)
secret=
callerid="GN"

[home](default)
secret=
username=home

I'm using Zoiper on Android for gn,

home is from the home machine, which registers with the server:

register => home:@

[home]
type=friend
insecure=port,invite
secret=; same secret as on server
context=incoming
host=

I'm wondering if it's a result of the amazon ec2 firewall (not 
iptables). I may need to open up those lower udp ports. Maybe the amazon 
firewall doesn't use ctstate; it may block any port not explicitly 
opened even if a connection is established.


Thanks for the help.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 08:04 AM, Julian Beach wrote:

Hello Sean,

Sunday, September 8, 2013, 11:25:24 PM, you wrote:


The problem is that once a phone has used the server, no other phone can
use it. The servers sees all the phones as having the same ip address
(though different ports).


This  sounds  like  the  Peer v Friend problem I have had in the past.
Try  setting  user=friend which will match on the username and not IP
address.  I  found  that asterisk was matching to the first account in
the  list  in  IAX.CONF  and  authentication  was  then failing (or in
the case of incoming calls, ending up in the wrong context).

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

Julian



Thanks for the response.

On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other 
peers are assigned 1026, 1027 and 1028. How are these ports assigned?


sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Joshua Colp]

Sean Darcy wrote:


On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other
peers are assigned 1026, 1027 and 1028. How are these ports assigned?


The actual configuration entries (minus password) for each one involved 
would be useful... if you aren't being explicit with what username to 
use for outgoing authentication then stuff like this can happen.


--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Eric Wieling]

They are assigned by the router doing the NAT translations.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 10:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

On 09/09/2013 08:04 AM, Julian Beach wrote:
> Hello Sean,
>
> Sunday, September 8, 2013, 11:25:24 PM, you wrote:
>
>> The problem is that once a phone has used the server, no other phone 
>> can use it. The servers sees all the phones as having the same ip 
>> address (though different ports).
>
> This  sounds  like  the  Peer v Friend problem I have had in the past.
> Try  setting  user=friend which will match on the username and not IP 
> address.  I  found  that asterisk was matching to the first account in 
> the  list  in  IAX.CONF  and  authentication  was  then failing (or in 
> the case of incoming calls, ending up in the wrong context).
>
> http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
>
> Julian
>
>
Thanks for the response.

On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other peers 
are assigned 1026, 1027 and 1028. How are these ports assigned?

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Julian Beach]

Hello Sean,

Sunday, September 8, 2013, 11:25:24 PM, you wrote:

> The problem is that once a phone has used the server, no other phone can
> use it. The servers sees all the phones as having the same ip address 
> (though different ports).

This  sounds  like  the  Peer v Friend problem I have had in the past.
Try  setting  user=friend which will match on the username and not IP
address.  I  found  that asterisk was matching to the first account in
the  list  in  IAX.CONF  and  authentication  was  then failing (or in
the case of incoming calls, ending up in the wrong context).

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

Julian


-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] IAX2 netsock error with name resolution

[Gopalakrishnan N]

After changing my dialplan as suggested, there is no socket error, but
getting Busy/Congested, and the call is hanging up, let me check that
part...

Earlier my dialplan was,
;exten => _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30)

and I changed like this exten => _2XXX,1,Dial(${MANIAX}/${EXTEN},30)

whether the SIP matters?

And now since its a SIP extension in other side, am getting failed because
the extension is not able to find.


Regards.


On Sun, Jun 23, 2013 at 5:22 PM, Alec Davis  wrote:

> 
> > -- Executing [2001@Test:1] Dial("SIP/4090-0005",
> "SIP/2001@IAX2/IND-MAN,30") in new stack
> > [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
> sip_request_call: Conflicting extension values given. Using '2001' and not
> 'IND-MAN'
> >   == Using SIP RTP CoS mark 5
> > [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
> ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
> in name resolution
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
> No such host: IAX2
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
> Subscriber absent)
> >   == Everyone is busy/congested at this time (1:0/0/1)
>
> Try this syntax Dial(IAX2/IND-MAN/2001,30)
> Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
> and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
> SIP/2001
>
> Alec Davis
>
>
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Re: [asterisk-users] IAX2 netsock error with name resolution

[Alec Davis]

> -- Executing [2001@Test:1] Dial("SIP/4090-0005",
"SIP/2001@IAX2/IND-MAN,30") in new stack
> [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
sip_request_call: Conflicting extension values given. Using '2001' and not
'IND-MAN'
>   == Using SIP RTP CoS mark 5
> [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
in name resolution
> [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
No such host: IAX2 
> [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
>   == Everyone is busy/congested at this time (1:0/0/1)

Try this syntax Dial(IAX2/IND-MAN/2001,30)
Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
SIP/2001

Alec Davis


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Re: [asterisk-users] IAX2 support of video

[Carlos Chavez]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 1/7/13 6:53 PM, Jerry Geis wrote:
>> 
>> According to this: 
>> https://wiki.asterisk.org/wiki/display/AST/Video+Telephony yes.
>> 
>> 
>> 
> I have a local server with two video phones - running SIP to each
> phone. Works. Then I have an IAX2 connection from that local
> machine to another machine. then a SIP connection from that machine
> to another machine where the same model video phone is in use. A
> call to that phone does not show video only audio.
> 
> All machines have in sip.conf:videosupport=yes
> 
> Is there something else to get SIP/IAX2/SIP video call to work?
> 
> Thanks
> 
> Jerry
> 
Make sure that your iax.conf entries for the link between servers
also allows the video codecs.


- -- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Comment: GPGTools - http://gpgtools.org
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=wUsG
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Re: [asterisk-users] IAX2 support of video

[Hans Witvliet]

On Tue, 2013-01-08 at 08:21 -0600, Danny Nicholas wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry
> Geis
> Sent: Monday, January 07, 2013 6:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX2 support of video
> 
>  
> 
>  
>  
> According to this:
> https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
> yes.
>  
>  
> 
>  
> 
> 
> I have a local server with two video phones - running SIP to each
> phone. Works.
> Then I have an IAX2 connection from that local machine to another
> machine.
> then a SIP connection from that machine to another machine where the
> same model
> video phone is in use. A call to that phone does not show video only
> audio.
> 
> All machines have in sip.conf:videosupport=yes
> 
> Is there something else to get SIP/IAX2/SIP video call to work?
> 
> Thanks
> 
> Jerry
> 
>  
> 
> Make sure you have the H.26X codec enabled at all points.
> 
> 
Video is hard, but to make life easier, it is handy to add an extension
that does the echo-function (after an optional announcement)
That takes video-codec-mismatch out of the equation, as you are talking
to yourself.

Other benefit is that there is always someone that will answer the
phone ;-) And won't complain doing video-cakk when dialing at 3am.

hans



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Re: [asterisk-users] IAX2 support of video

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 support of video

 

 
 
According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
yes.
 
 

 

I have a local server with two video phones - running SIP to each phone.
Works.
Then I have an IAX2 connection from that local machine to another machine.
then a SIP connection from that machine to another machine where the same
model
video phone is in use. A call to that phone does not show video only audio.

All machines have in sip.conf:videosupport=yes

Is there something else to get SIP/IAX2/SIP video call to work?

Thanks

Jerry

 

Make sure you have the H.26X codec enabled at all points.

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Re: [asterisk-users] IAX2 support of video

[Jerry Geis]

According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
yes.



I have a local server with two video phones - running SIP to each phone. 
Works.

Then I have an IAX2 connection from that local machine to another machine.
then a SIP connection from that machine to another machine where the 
same model

video phone is in use. A call to that phone does not show video only audio.

All machines have in sip.conf:videosupport=yes

Is there something else to get SIP/IAX2/SIP video call to work?

Thanks

Jerry
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Re: [asterisk-users] IAX2 support of video

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 support of video

Does IAX2 support a video call ?

According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
yes.


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Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

[Dave Platt]

>>  Here's where I am baffled and I am hoping someone with intricate
>> knowledge of this implementation may be able to explain it to me. What
>> we had to do to get this working was to set the host= parameter to the
>> respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
>> 172.10.1.2 in his case. Calls flow normally now and we cannot understand
>> how or why. I would have assumed with a destination of either LAN as
>> defined by the routing table it would have left out on the OpenVPN
>> connection by default, and what's even more strange is that IAX is the
>> only protocol that does not appear to function as intended.


> My guess is asterisk is replying using the tunnel ip address which your 
> original box won't accept unless you actually sent to that address. Thats 
> what I see on our remote openvpn tunnels. If you want to know whats going on 
> use tcpdump to check packets through the tunnel. 

Yes, I've seen this same problem.  It has two possible solutions.

The reason for the problem is this:  IAX2 (the Asterisk
implementation, at least) depends on the "source" address in the
UDP packet it receives, to know which connection the packet
is part of.  When it talks to a peer, it expects to see the
packets arrive from the peer with a source address which
matches what it understands the peer's address to be.  Packets
arriving from "unknown" addresses, are simply dropped on the
floor (considered to be misrouted, misconfigured, or forged,
I think).

Normally, the Asterisk IAX2 implementation does not
bind itself to a single network interface.  It will
receive UDP packets to the IAX2 port, arriving from
any interface.

And, when it sends an IAX2 UDP packet, it simply sends
it out through the socket which is bound to the
"any interface" port.

Because the socket isn't bound to a specific interface,
it doesn't have a specific IP address associated with it.
The Linux kernel chooses an IP address to put into the
UDP packet "source" field, and the one it chooses is the
IP address of the interface on which it is transmitting
the packet.

In the scenario that's being described here, an address
result mismatches.  Each system is transmitting UDP packets
*to* the "primary" or "official" or "public" interface on
its peer... and these packets are being transmitted by
the Linux kernel on the OpenVPN interface, and are being
given the system's OpenVPN tunnel endpoint address.  In each
case, when the packet arrives at the peer, the Asterisk IAX2
stack receives the packets, finds that it has no known peer
at the tunnel IP address and no IAX2 session set up for this
address, and discards the packet.

There are, I believe, two solutions which don't involve
modifying the IAX2 code in Asterisk.  Both work equally
well, as far as I know.

One approach is the one you've taken - tell each system to
"talk to" its peer's OpenVPN tunnel endpoint address, rather
than to the "primary" address.  This eliminates the IP address
mismatch.  This approach works fine if both systems are connected
only via this OpenVPN tunnel, and always have the same OpenVPN
addresses.

The other approach is to configure each system to bind its
IAX2 port *only* to one IP interface (usually the public one),
to ensure that each peer knows how to reach its peer's
public IP address (either directly, or via a route though the
OpenVPN tunnel), and to tell each system to speak IAX2 to its
peer's public IP address.

In this case, since the Asterisk socket is bound to a specific
interface, all packets sent through that socket will have
the bound interface's IP address in its "source" field, and
(once again) the address mismatch is eliminated.

This second approach is preferable for "road warrior"
configurations in which you might sometimes be using the
OpenVPN tunnel, and sometimes not (e.g. a laptop or tablet
IAX2 client which is sometimes on the corporate LAN and
sometimes out on the Internet using OpenVPN).



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Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

[Steve Totaro]

On Sun, Dec 9, 2012 at 2:54 PM, Stephen Brown  wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> So a friend of mine and I setup a static key based point to point
> OpenVPN connection from my box to his for the express intent of carrying
> IAX traffic encrypted.
>
> His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His
> PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an
> existing working IAX trunk in place prior to the VPN, and after we
> brought the VPN up we set the host= parameter within Asterisk
> accordingly on each end to match the local IP's and discovered it did
> not work. The trunk remained in an UNKNOWN status on each end, even
> though we could ping each box locally, SSH, and even SIP worked.
>
> Here's where I am baffled and I am hoping someone with intricate
> knowledge of this implementation may be able to explain it to me. What
> we had to do to get this working was to set the host= parameter to the
> respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
> 172.10.1.2 in his case. Calls flow normally now and we cannot understand
> how or why. I would have assumed with a destination of either LAN as
> defined by the routing table it would have left out on the OpenVPN
> connection by default, and what's even more strange is that IAX is the
> only protocol that does not appear to function as intended.
>
> Any takers? :)
>
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v2.0.17 (MingW32)
>
> iEYEARECAAYFAlDE7GcACgkQ3sJXNEncx7is9QCcCciMYFJ7ZXjYxuHC2EYD0PZY
> waAAniNNx8GuC5To7ajlGR5sYs3yftFK
> =lcWJ
> -END PGP SIGNATURE-
>
>

First, not so much of an answer but more of a question.  Why use IAX2
in your scenario?  SIP would seem to be very logical in this case if
you already tested it and it works.

IAX2 really only has merits where NAT and multiple ports are an issue.
 It has been known to create many problems and headaches.

Since OpenVPN negates the multiple ports over the web, and NAT isn't a
problem from what you have stated, why even bother with IAX2?

To cleanly solve your issue, create an OpenVPN tunnel directly between
the boxen with the same IP/subnet scheme.  That is what I would do, as
each tunnel is a "subinterface" of sorts, there is no need to keep the
addressing scheme of your LANs.  SIP and IAX2 should both work for you
(I still suggest SIP).  Creating a separate subnet for your OpenVPN
connection will arguably also add a bit of security between networks.

What does your IPtables look like?  Maybe you are blocking IAX?  Turn
of debugging and post verbose.

Thanks,
Steve T

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Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

[Duncan Turnbull]

On 10/12/2012, at 8:54 AM, Stephen Brown  wrote:

> 
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> So a friend of mine and I setup a static key based point to point
> OpenVPN connection from my box to his for the express intent of carrying
> IAX traffic encrypted.
> 
> His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His
> PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an
> existing working IAX trunk in place prior to the VPN, and after we
> brought the VPN up we set the host= parameter within Asterisk
> accordingly on each end to match the local IP's and discovered it did
> not work. The trunk remained in an UNKNOWN status on each end, even
> though we could ping each box locally, SSH, and even SIP worked.
> 
> Here's where I am baffled and I am hoping someone with intricate
> knowledge of this implementation may be able to explain it to me. What
> we had to do to get this working was to set the host= parameter to the
> respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
> 172.10.1.2 in his case. Calls flow normally now and we cannot understand
My guess is asterisk is replying using the tunnel ip address which your 
original box won't accept unless you actually sent to that address. Thats what 
I see on our remote openvpn tunnels. If you want to know whats going on use 
tcpdump to check packets through the tunnel. 

> how or why. I would have assumed with a destination of either LAN as

> defined by the routing table it would have left out on the OpenVPN
> connection by default, and what's even more strange is that IAX is the
> only protocol that does not appear to function as intended.
> 
> Any takers? :)
> 
> 
> -BEGIN PGP SIGNATURE-
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> =lcWJ
> -END PGP SIGNATURE-
> 
> 
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Re: [asterisk-users] iax2-provision.c:266 iax_provision_version: ast_db_get failed to retrieve iax/provisioning/cach

[Richard Mudgett]

> > I've tested asterisk 1.8.17.0 and I'm still getting the repeated
> > error message on the command line:
> > 
> > iax2-provision.c:266 iax_provision_version: ast_db_get failed to
> > retrieve iax/provisioning/cach
> 
> Are you out of disk space?  I would only expect to see that message
> once
> since it looks like the code attempts to correct the problem.

Never mind.  I was looking at code that already had the patch for
ASTERISK-20337 included.

Richard

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Re: [asterisk-users] iax2-provision.c:266 iax_provision_version: ast_db_get failed to retrieve iax/provisioning/cach

[Richard Mudgett]

> I've tested asterisk 1.8.17.0 and I'm still getting the repeated
> error message on the command line:
> 
> iax2-provision.c:266 iax_provision_version: ast_db_get failed to
> retrieve iax/provisioning/cach

Are you out of disk space?  I would only expect to see that message once
since it looks like the code attempts to correct the problem.

Richard

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Re: [asterisk-users] iax2 trunks between asterisk servers

[Stephen Collier]

Doug,

Thanks, that answers my question I will reuse the macro I'm using with
an Avaya connection and CONNECTEDLINE(). Pity I was hoping iax2 would
transfer callee id.

Cheers
Stephen


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Re: [asterisk-users] iax2 trunks between asterisk servers

[Doug Lytle]

Stephen Collier wrote:

Any ideas or suggestions appreciated.


We keep an mysql database of all extensions (Fax2Email) that I use to do 
a lookup against the destination extension and then set the phone to 
display the name.


Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] IAX2 Registered OK without IP

[Duncan Turnbull]

On 27/07/2012, at 8:16 AM, Alejandro Imass  wrote:

> On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass  wrote:
>> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
>> we upgraded to 1.8.13.1 and we have much the same problem although after
>> the upgrade I don't seem to find any cases where the qualify value is
>> OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
>> is the same: the extension becomes non-reachable pretty quickly.
>> 
> 
Can you confirm whether you have a firewall between the phones or not? And also 
using tcpdump and IAX debug what packets you are seeing

This is a network problem and something is disrupting the return packets so you 
need to see where it's occurring 

>From the cli use the iax2 set debug command and watch what's happening. Are 
>the packets being returned? If they aren't check with tcpdump to see if they 
>are at least getting to your interface

Do the same at the other end to work out whats missing


> I stand corrected. It's EXACTLY the same behavior as 1.4.29, The
> Status shows OK (XX ms) and the IP is "(null)"
> 
>> IMHO this is indicating that the qualify settings are being ignored,
> 
> I've been experimenting with qualifyfreqok and qualifyfreqnotok and
> just by specifiying _any_ values for these parameters it makes matters
> much worse.
> 
>> and the only workaround has been lowering the re-register time to
>> sometimes as low as 3 seconds. Even though several docs say that the
> 
> A re-registration every 3 seconds seems to do the trick but why can't
> qualify keep the connection alive??
> 
>> qualify values can also act as a keep alive, it's not working for us
>> and I still have to reduce the register time to very low values
>> because qualifyfreqok and qualifyfreqnotok don't see to be doing
>> anything...
>> 
> 
> Stand corrected: they just make the problem worse.
> 
>> Any clues??
> 
> I get the feeling that IAX extensions are definitively not very
> popular even though most of the older concerns about IAX should be
> gone by now. We really want to support and keep investing into making
> IAX work for us but the little support we get here is really
> discouraging.
> 
> Best,
> 
> -- 
> Alejandro Imass
> 
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Re: [asterisk-users] IAX2 Registered OK without IP

[Alejandro Imass]

On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass  wrote:
> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
> we upgraded to 1.8.13.1 and we have much the same problem although after
> the upgrade I don't seem to find any cases where the qualify value is
> OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
> is the same: the extension becomes non-reachable pretty quickly.
>

I stand corrected. It's EXACTLY the same behavior as 1.4.29, The
Status shows OK (XX ms) and the IP is "(null)"

> IMHO this is indicating that the qualify settings are being ignored,

I've been experimenting with qualifyfreqok and qualifyfreqnotok and
just by specifiying _any_ values for these parameters it makes matters
much worse.

> and the only workaround has been lowering the re-register time to
> sometimes as low as 3 seconds. Even though several docs say that the

A re-registration every 3 seconds seems to do the trick but why can't
qualify keep the connection alive??

> qualify values can also act as a keep alive, it's not working for us
> and I still have to reduce the register time to very low values
> because qualifyfreqok and qualifyfreqnotok don't see to be doing
> anything...
>

Stand corrected: they just make the problem worse.

> Any clues??

I get the feeling that IAX extensions are definitively not very
popular even though most of the older concerns about IAX should be
gone by now. We really want to support and keep investing into making
IAX work for us but the little support we get here is really
discouraging.

Best,

-- 
Alejandro Imass

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Re: [asterisk-users] IAX2 Registered OK without IP

[Duncan Turnbull]

On 25/07/2012, at 7:54 AM, Alejandro Imass wrote:

> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
>> This has come up before on the list and archives but I don't seem to
>> find a solution for this. On just a few nodes we have this situation
>> where we see the IP disappear from the CLI iax2 show peers list but
>> the status shows OK:
>> 
>> 3012/3012(Unspecified)   (D)  255.255.255.255  0 OK (89 
>> ms)
>> 
Not sure how relevant it is but IAX can appear to be working on one links 
available in only one direction. If a firewall is blocking one end then one end 
may appear fine, and the other disconnected. 

It might be a NAT issue causing the blocking where registering opens the NAT 
tables but they time out - this used to happen on older home firewalls

Or it could be something else quite odd
- can you confirm you have network connectivity to the end devices? 
- does it happen to devices on the local lan or just via firewall

I doubt very much its the quality settings, they are just reflecting whats 
being seen rather than the cause of the issue. When you turn on IAX debug do 
you see the qualify packets and replies  (pings and pongs) in asterisk or are 
they missing? If so then also use tcpdump to see if you are actually getting 
the packets you would expect at your network card to figure out if its internal 
to the server or related to the network

Good luck

>> How can the status be OK a few milliseconds ago and have no IP ?? The
>> strange thing is that the IP does show up once in a while and then
>> disappears once again but the OK is always there.
>> 
>> Asterisk 1.4.29 running on FreeBSD 7.0-RELEASE
> 
> OK, so I guess nobody answered because we had an old version. Well, we
> upgraded to 1.8.13.1 and we have much the same problem although after
> the upgrade I don't seem to find any cases where the qualify value is
> OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
> is the same: the extension becomes non-reachable pretty quickly.
> 
> IMHO this is indicating that the qualify settings are being ignored,
> and the only workaround has been lowering the re-register time to
> sometimes as low as 3 seconds. Even though several docs say that the
> qualify values can also act as a keep alive, it's not working for us
> and I still have to reduce the register time to very low values
> because qualifyfreqok and qualifyfreqnotok don't see to be doing
> anything...
> 
> Any clues??
> 
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Re: [asterisk-users] IAX2 Registered OK without IP

[Alejandro Imass]

On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
> This has come up before on the list and archives but I don't seem to
> find a solution for this. On just a few nodes we have this situation
> where we see the IP disappear from the CLI iax2 show peers list but
> the status shows OK:
>
> 3012/3012(Unspecified)   (D)  255.255.255.255  0 OK (89 
> ms)
>
> How can the status be OK a few milliseconds ago and have no IP ?? The
> strange thing is that the IP does show up once in a while and then
> disappears once again but the OK is always there.
>
> Asterisk 1.4.29 running on FreeBSD 7.0-RELEASE

OK, so I guess nobody answered because we had an old version. Well, we
upgraded to 1.8.13.1 and we have much the same problem although after
the upgrade I don't seem to find any cases where the qualify value is
OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
is the same: the extension becomes non-reachable pretty quickly.

IMHO this is indicating that the qualify settings are being ignored,
and the only workaround has been lowering the re-register time to
sometimes as low as 3 seconds. Even though several docs say that the
qualify values can also act as a keep alive, it's not working for us
and I still have to reduce the register time to very low values
because qualifyfreqok and qualifyfreqnotok don't see to be doing
anything...

Any clues??

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Re: [asterisk-users] IAX2 passing back and forth variables

[Larry Moore]

On 23/05/2012 10:46 AM, Ruddy Gbaguidi wrote:


I cannot find it

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny 
Nicholas

*Sent:* 2012-05-21 10:25
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] IAX2 passing back and forth variables

There was a nice thread on this back in April.




Perhaps it is the thread which started on the 15th of April with the 
subject line


Set variables from one asterisk ta a second.





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Re: [asterisk-users] IAX2 passing back and forth variables

[Ruddy Gbaguidi]

I cannot find it

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 2012-05-21 10:25 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

There was a nice thread on this back in April.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Monday, May 21, 2012 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

No one have an idea ?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Sorry, the dialplan is really on server B

exten => s,n,Set(IAXVAR(TESTVAR2)=efgh) 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45 
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything.  You would need some sort of Set(IAXVAR(TESTVAR2)=.)

 

Noah

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables

 

Hi all,

I have two asterisk servers A and B.

And I would like from A, dial to B passing some IAX variables.

Then B handles the calls, setup some other variables that become available
to A which can continue.

So far, I have used IAXVAR function.

It works when sending call from A to B

But variables setup on B are not available on A.

 

Any idea how I can do it ?

 

Here are my dialplans.

+++

SERVER A

+++

[contextA]

exten => s,1,Set(IAXVAR(TESTVAR1)=abcd)

exten => s,n,Dial(IAX2/serverb/s,30,g)

exten => s,n,Noop(  The out variable is : ${IAXVAR(TESTVAR2)}   )  ; <
Does not work

 

 

+++

SERVER B

+++

[contextB]

exten => s,1,Noop( ${IAXVAR(TESTVAR1)} )   <- Does work

exten => s,n,Set(IAXVAR(TESTVAR2)) 

exten => s,n,Hangup

 

 

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Re: [asterisk-users] IAX2 passing back and forth variables

[Danny Nicholas]

There was a nice thread on this back in April.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Monday, May 21, 2012 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

No one have an idea ?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Sorry, the dialplan is really on server B

exten => s,n,Set(IAXVAR(TESTVAR2)=efgh) 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45 
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything.  You would need some sort of Set(IAXVAR(TESTVAR2)=.)

 

Noah

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables

 

Hi all,

I have two asterisk servers A and B.

And I would like from A, dial to B passing some IAX variables.

Then B handles the calls, setup some other variables that become available
to A which can continue.

So far, I have used IAXVAR function.

It works when sending call from A to B

But variables setup on B are not available on A.

 

Any idea how I can do it ?

 

Here are my dialplans.

+++

SERVER A

+++

[contextA]

exten => s,1,Set(IAXVAR(TESTVAR1)=abcd)

exten => s,n,Dial(IAX2/serverb/s,30,g)

exten => s,n,Noop(  The out variable is : ${IAXVAR(TESTVAR2)}   )  ; <
Does not work

 

 

+++

SERVER B

+++

[contextB]

exten => s,1,Noop( ${IAXVAR(TESTVAR1)} )   <- Does work

exten => s,n,Set(IAXVAR(TESTVAR2)) 

exten => s,n,Hangup

 

 

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Re: [asterisk-users] IAX2 passing back and forth variables

[Ruddy Gbaguidi]

No one have an idea ?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Sorry, the dialplan is really on server B

exten => s,n,Set(IAXVAR(TESTVAR2)=efgh) 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45 
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything.  You would need some sort of Set(IAXVAR(TESTVAR2)=.)

 

Noah

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables

 

Hi all,

I have two asterisk servers A and B.

And I would like from A, dial to B passing some IAX variables.

Then B handles the calls, setup some other variables that become available
to A which can continue.

So far, I have used IAXVAR function.

It works when sending call from A to B

But variables setup on B are not available on A.

 

Any idea how I can do it ?

 

Here are my dialplans.

+++

SERVER A

+++

[contextA]

exten => s,1,Set(IAXVAR(TESTVAR1)=abcd)

exten => s,n,Dial(IAX2/serverb/s,30,g)

exten => s,n,Noop(  The out variable is : ${IAXVAR(TESTVAR2)}   )  ; <
Does not work

 

 

+++

SERVER B

+++

[contextB]

exten => s,1,Noop( ${IAXVAR(TESTVAR1)} )   <- Does work

exten => s,n,Set(IAXVAR(TESTVAR2)) 

exten => s,n,Hangup

 

 

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Re: [asterisk-users] IAX2 passing back and forth variables

[Ruddy Gbaguidi]

Sorry, the dialplan is really on server B

exten => s,n,Set(IAXVAR(TESTVAR2)=efgh) 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45 
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables

 

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything.  You would need some sort of Set(IAXVAR(TESTVAR2)=.)

 

Noah

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables

 

Hi all,

I have two asterisk servers A and B.

And I would like from A, dial to B passing some IAX variables.

Then B handles the calls, setup some other variables that become available
to A which can continue.

So far, I have used IAXVAR function.

It works when sending call from A to B

But variables setup on B are not available on A.

 

Any idea how I can do it ?

 

Here are my dialplans.

+++

SERVER A

+++

[contextA]

exten => s,1,Set(IAXVAR(TESTVAR1)=abcd)

exten => s,n,Dial(IAX2/serverb/s,30,g)

exten => s,n,Noop(  The out variable is : ${IAXVAR(TESTVAR2)}   )  ; <
Does not work

 

 

+++

SERVER B

+++

[contextB]

exten => s,1,Noop( ${IAXVAR(TESTVAR1)} )   <- Does work

exten => s,n,Set(IAXVAR(TESTVAR2)) 

exten => s,n,Hangup

 

 

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Re: [asterisk-users] IAX2 passing back and forth variables

[Noah Engelberth]

Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2 to 
anything.  You would need some sort of Set(IAXVAR(TESTVAR2)=...)

Noah

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables

Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available to A 
which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.

Any idea how I can do it ?

Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten => s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten => s,n,Dial(IAX2/serverb/s,30,g)
exten => s,n,Noop(  The out variable is : ${IAXVAR(TESTVAR2)}   )  ; < Does 
not work


+++
SERVER B
+++
[contextB]
exten => s,1,Noop( ${IAXVAR(TESTVAR1)} )   <- Does work
exten => s,n,Set(IAXVAR(TESTVAR2))
exten => s,n,Hangup


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Re: [asterisk-users] IAX2 woes

[Carlos Rojas]

Hello

Asterisk only says that the iax2 channel don't work maybe you look the
iax.conf. you trunk. Is iax I think

Regards
On Dec 29, 2011 6:49 AM, "--[ UxBoD ]--"  wrote:

> Hello all,
>
> I attempted to make a couple of outbound calls this morning and always got
> the busy tone.  I checked the Asterisk console and was greeted with:
>
> [Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable
> to create channel of type 'IAX2' (cause 20 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
>
> I proceeded to restart Asterisk and dialed the same number again and it
> worked without fault. What could cause this type of error and is there any
> way to auto-remediate when it does arise ?
>
> voip*CLI> core show version
> Asterisk 10.0.0 built by root @ voip.my.server on a x86_64 running Linux
> on 2011-12-19 16:16:46 UTC
> --
> Thanks, Phil
>
>
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Re: [asterisk-users] IAX2 availability testing

[Danny Nicholas]

ChanisAvail?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent: Wednesday, November 09, 2011 9:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 availability testing

Hi folks,

What methods are available for testing IAX2 service availability? I know
about "iax2 show peers" and "iax2 show registry", but I'd like some
alternatives.

Tcpdump shows a little more about what's going on, but a handy test using
nmap doesn't seem to work anymore (see
http://shearer.org/UDP_Reachability_Testing).

Any suggestions would be appreciated.

Cheers,

Jaap

PS -- My systems run Debian squeeze with Asterisk 1.6.2.9.

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Re: [asterisk-users] iax2 Max retries exceeded to host

[satish patel]

campbx1*CLI> iax2 show netstats
    LOCAL -   
REMOTE 
Channel   RTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit  Del  Lost 
  %  Drop  OOO  Kpkts FirstMsgLastMsg
IAX2/orasebcam-612 83   -10-1  -1 0   -1  00   40 0 
  0 00  0 Tx:NEW  Tx:LAGRQ  
IAX2/7504-1407204   -10-1  -1 0   -120200 0 
  0 00  0 Rx:NEW  Tx:ACK
IAX2/orasebcam-3360   104   -10-1  -1 0   -1  50   40 0 
  0 00  0 Rx:NEW  Rx:ACK
IAX2/orasebcam-828784   -10-1  -1 0   -12020   40 0 
  0 00  0 Tx:NEW  Rx:ACK
IAX2/7504-15510   178   -10-1  -1 0   -1  200 0 
  0 00  0 Rx:NEW  Tx:ACK
5 active IAX channels


From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 10 May 2011 19:27:26 +
Subject: [asterisk-users] iax2 Max retries exceeded to host










We have IAX2 peer between two asterisk and I am getting following error 
following IAX2 WARNING. IAX calling is functional 

[May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 11, ts=3030332, seqno=211)
[May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 11, ts=3040332, seqno=212)
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: 
Registration from '' failed for 
'172.30.245.85:5060' - No matching peer found
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: 
Registration from '' failed for 
'172.30.245.85:5060' - No matching peer found
[May 10 15:23:49] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 2, ts=3045385, seqno=213)
[May 10 15:23:54] WARNING[2054]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 11, ts=3050332, seqno=214)
[May 10 15:24:04] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 11, ts=3060332, seqno=215)
[May 10 15:24:10] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 2, ts=3066385, seqno=216)
[May 10 15:24:14] WARNING[2051]: chan_iax2.c:3487 __attempt_transmit: Max 
retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, 
subclass = 11, ts=3070332, seqno=217)

  

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Re: [asterisk-users] iax2 issue in asterisk

[satish patel]

Awesome! 

root@:~# cat /etc/asterisk/iax.conf | grep requirecalltoken
; By setting 'requirecalltoken=no', call token validation becomes optional for
; that peer/user.  By setting 'requirecalltoken=auto', call token validation 
; can require it from this peer.  So, requirecalltoken is internally set to yes.
; requirecalltoken may only be used in peer/user/friend definitions,
; By default, 'requirecalltoken=yes'.

requirecalltoken=no

Also there was an other issue host=dynamic i set to host=x.x.x.x and it works!


> Date: Mon, 9 May 2011 18:45:05 +0100
> From: gordon+aster...@drogon.net
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] iax2 issue in asterisk
> 
> On Mon, 9 May 2011, satish patel wrote:
> 
> >
> > Hey guys!
> >
> > I have issue between iax vs iax2 following is my setup
> >
> > asterisk-1.2 <--IAX>Asterisk-1.8
> >
> > I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8
> 
> 
> Might you be missing
> 
>requirecalltoken=no
> 
> in iax.conf in the 1.8 system for calls originating from the 1.2 system?
> 
> Gordon
> 
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Re: [asterisk-users] iax2 issue in asterisk

[Gordon Henderson]

On Mon, 9 May 2011, satish patel wrote:



Hey guys!

I have issue between iax vs iax2 following is my setup

asterisk-1.2 <--IAX>Asterisk-1.8

I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8



Might you be missing

  requirecalltoken=no

in iax.conf in the 1.8 system for calls originating from the 1.2 system?

Gordon

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Re: [asterisk-users] IAX2 codec selection and video

[Pezhman Lali]

check this url, let me know if any problem

http://www.voip-info.org/wiki/view/Asterisk+video


 best


On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies  wrote:

> Hi,
>
> Can anyone let me know how I can enable video (h.263) on SIP, but if a
> video call is passed over IAX, it will remove the video and pass the
> audio only.
>
> What I tried was:
>
> SIP - videosupport=yes
>  - disallow=all
>  - allow=alaw
>  - allow=h263
>
> IAX - disallow=all
>  - allow=alaw
>
>
> What appears to occur is that the SIP call negotiates h263 video, and
> when passed over IAX, the h263  frames are passed, and are also
> accepted at the far end which also does not have a video codec
> allowed. Should that be happening? This is with 1.6.2.18-rc1. Am I
> missing a setting somewhere?
>
> Thanks,
> Steve
>
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Re: [asterisk-users] IAX2/0.0.29.199

[Satish Patel]

I grab via svn client and source you gave me.

Can you fix original brach ?

--
Sent from my iPhone

On Apr 10, 2011, at 11:51 AM, Paul Belanger   
wrote:



On 11-04-10 09:14 AM, Tzafrir Cohen wrote:

On Fri, Apr 08, 2011 at 06:10:21PM +, satish patel wrote:



I tried to compile your version and got bunch of error on "make"  
and it failed to compile.


root@satish-desktop:/home/satish/issue18183# make


How did you get that code?

It is from a branch I created a few months back, and I have not  
looked at it in a while.  That said, there maybe issues with it.


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Re: [asterisk-users] IAX2/0.0.29.199

[Paul Belanger]

On 11-04-10 09:14 AM, Tzafrir Cohen wrote:

On Fri, Apr 08, 2011 at 06:10:21PM +, satish patel wrote:



I tried to compile your version and got bunch of error on "make" and it failed 
to compile.

root@satish-desktop:/home/satish/issue18183# make


How did you get that code?

It is from a branch I created a few months back, and I have not looked 
at it in a while.  That said, there maybe issues with it.


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Re: [asterisk-users] IAX2/0.0.29.199

[Tzafrir Cohen]

On Fri, Apr 08, 2011 at 06:10:21PM +, satish patel wrote:
> 
> 
> I tried to compile your version and got bunch of error on "make" and it 
> failed to compile. 
> 
> root@satish-desktop:/home/satish/issue18183# make

How did you get that code?

>[CC] chan_iax2.c -> chan_iax2.o
> chan_iax2.c: In function âsocket_processâ:
> chan_iax2.c:11533: error: invalid storage class for function 
> âiax2_process_thread_cleanupâ
> chan_iax2.c:11532: warning: no previous prototype for 
> âiax2_process_thread_cleanupâ
> chan_iax2.c:11544: error: invalid storage class for function 
> âiax2_process_threadâ
> chan_iax2.c:11543: warning: no previous prototype for âiax2_process_threadâ
> chan_iax2.c:11683: error: invalid storage class for function 
> âiax2_do_registerâ
> chan_iax2.c:11682: warning: no previous prototype for âiax2_do_registerâ
> chan_iax2.c:11744: error: invalid storage class for function âiax2_provisionâ
> chan_iax2.c:11743: warning: no previous prototype for âiax2_provisionâ
> chan_iax2.c:11796: error: invalid storage class for function âiax2_prov_appâ
> chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_appâ
> chan_iax2.c:11825: error: invalid storage class for function 
> âhandle_cli_iax2_provisionâ
> chan_iax2.c:11824: warning: no previous prototype for 
> âhandle_cli_iax2_provisionâ
> chan_iax2.c:11864: error: invalid storage class for function 
> â__iax2_poke_noanswerâ
> chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_noanswerâ
> chan_iax2.c:11887: error: invalid storage class for function 
> âiax2_poke_noanswerâ
> ...
> ...
> ...
> chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ
> chan_iax2.c:14723: error: invalid storage class for function â__unreg_moduleâ
> chan_iax2.c:14723: warning: no previous prototype for â__unreg_moduleâ
> chan_iax2.c:14723: error: expected declaration or statement at end of input
> chan_iax2.c:14723: warning: unused variable âast_module_infoâ
> make[1]: *** [chan_iax2.o] Error 1
> make: *** [channels] Error 2
> root@satish-desktop:/home/satish/issue18183#

-- 
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] IAX2/0.0.29.199

[Satish Patel]

Bump up! Please help here

--
Sent from my iPhone

On Apr 8, 2011, at 2:10 PM, satish patel  wrote:



I tried to compile your version and got bunch of error on "make" and  
it failed to compile.


root@satish-desktop:/home/satish/issue18183# make
   [CC] chan_iax2.c -> chan_iax2.o
chan_iax2.c: In function âsocket_processâ:
chan_iax2.c:11533: error: invalid storage class for function âiax2_p 
rocess_thread_cleanupâ
chan_iax2.c:11532: warning: no previous prototype for âiax2_process_ 
thread_cleanupâ
chan_iax2.c:11544: error: invalid storage class for function âiax2_p 
rocess_threadâ
chan_iax2.c:11543: warning: no previous prototype for âiax2_process_ 
threadâ
chan_iax2.c:11683: error: invalid storage class for function âiax2_d 
o_registerâ
chan_iax2.c:11682: warning: no previous prototype for âiax2_do_regis 
terâ
chan_iax2.c:11744: error: invalid storage class for function âiax2_p 
rovisionâ
chan_iax2.c:11743: warning: no previous prototype for âiax2_provisio 
nâ
chan_iax2.c:11796: error: invalid storage class for function âiax2_p 
rov_appâ
chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_ap 
pâ
chan_iax2.c:11825: error: invalid storage class for function âhandle 
_cli_iax2_provisionâ
chan_iax2.c:11824: warning: no previous prototype for âhandle_cli_ia 
x2_provisionâ
chan_iax2.c:11864: error: invalid storage class for function â__iax2 
_poke_noanswerâ
chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_n 
oanswerâ
chan_iax2.c:11887: error: invalid storage class for function âiax2_p 
oke_noanswerâ

...
...
...
chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ
chan_iax2.c:14723: error: invalid storage class for function â__unre 
g_moduleâ
chan_iax2.c:14723: warning: no previous prototype for â__unreg_modul 
eâ
chan_iax2.c:14723: error: expected declaration or statement at end  
of input

chan_iax2.c:14723: warning: unused variable âast_module_infoâ
make[1]: *** [chan_iax2.o] Error 1
make: *** [channels] Error 2
root@satish-desktop:/home/satish/issue18183#





> Date: Fri, 8 Apr 2011 13:16:30 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>
> On 11-04-08 12:56 PM, Paul Belanger wrote:
> > On 11-04-08 11:55 AM, satish patel wrote:
> >>
> >> @Paul - many time i am gettting following SIP error when  
channel isn't
> >> available. I want to get rid on this revers thing. I tried all  
version

> >> 1.8.1,1.8.2,1.8.3 but not fix :(
> >>
> > Best you can do is collect a full debug[1] log and see when the  
issue is

> > introduced.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> >
> Do you mind trying the following branch[2]? Not sure if it will  
help,

> but I made some changes to chan_iax2 a few months ago.
>
> [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
>
> --
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> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

I tried to compile your version and got bunch of error on "make" and it failed 
to compile. 

root@satish-desktop:/home/satish/issue18183# make
   [CC] chan_iax2.c -> chan_iax2.o
chan_iax2.c: In function âsocket_processâ:
chan_iax2.c:11533: error: invalid storage class for function 
âiax2_process_thread_cleanupâ
chan_iax2.c:11532: warning: no previous prototype for 
âiax2_process_thread_cleanupâ
chan_iax2.c:11544: error: invalid storage class for function 
âiax2_process_threadâ
chan_iax2.c:11543: warning: no previous prototype for âiax2_process_threadâ
chan_iax2.c:11683: error: invalid storage class for function âiax2_do_registerâ
chan_iax2.c:11682: warning: no previous prototype for âiax2_do_registerâ
chan_iax2.c:11744: error: invalid storage class for function âiax2_provisionâ
chan_iax2.c:11743: warning: no previous prototype for âiax2_provisionâ
chan_iax2.c:11796: error: invalid storage class for function âiax2_prov_appâ
chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_appâ
chan_iax2.c:11825: error: invalid storage class for function 
âhandle_cli_iax2_provisionâ
chan_iax2.c:11824: warning: no previous prototype for 
âhandle_cli_iax2_provisionâ
chan_iax2.c:11864: error: invalid storage class for function 
â__iax2_poke_noanswerâ
chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_noanswerâ
chan_iax2.c:11887: error: invalid storage class for function 
âiax2_poke_noanswerâ
...
...
...
chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ
chan_iax2.c:14723: error: invalid storage class for function â__unreg_moduleâ
chan_iax2.c:14723: warning: no previous prototype for â__unreg_moduleâ
chan_iax2.c:14723: error: expected declaration or statement at end of input
chan_iax2.c:14723: warning: unused variable âast_module_infoâ
make[1]: *** [chan_iax2.o] Error 1
make: *** [channels] Error 2
root@satish-desktop:/home/satish/issue18183#





> Date: Fri, 8 Apr 2011 13:16:30 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 12:56 PM, Paul Belanger wrote:
> > On 11-04-08 11:55 AM, satish patel wrote:
> >>
> >> @Paul - many time i am gettting following SIP error when channel isn't
> >> available. I want to get rid on this revers thing. I tried all version
> >> 1.8.1,1.8.2,1.8.3 but not fix :(
> >>
> > Best you can do is collect a full debug[1] log and see when the issue is
> > introduced.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> >
> Do you mind trying the following branch[2]?  Not sure if it will help, 
> but I made some changes to chan_iax2 a few months ago.
> 
> [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

I have just compiled asterisk 1.6.x  and its working without any issue no error 
related revers lookup etc.. Look like there is some glitch in asterisk 1.8 :(  

-S


> Date: Fri, 8 Apr 2011 13:16:30 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 12:56 PM, Paul Belanger wrote:
> > On 11-04-08 11:55 AM, satish patel wrote:
> >>
> >> @Paul - many time i am gettting following SIP error when channel isn't
> >> available. I want to get rid on this revers thing. I tried all version
> >> 1.8.1,1.8.2,1.8.3 but not fix :(
> >>
> > Best you can do is collect a full debug[1] log and see when the issue is
> > introduced.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> >
> Do you mind trying the following branch[2]?  Not sure if it will help, 
> but I made some changes to chan_iax2 a few months ago.
> 
> [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

I have opened case here: https://issues.asterisk.org/view.php?id=19087 



> Date: Fri, 8 Apr 2011 13:16:30 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 12:56 PM, Paul Belanger wrote:
> > On 11-04-08 11:55 AM, satish patel wrote:
> >>
> >> @Paul - many time i am gettting following SIP error when channel isn't
> >> available. I want to get rid on this revers thing. I tried all version
> >> 1.8.1,1.8.2,1.8.3 but not fix :(
> >>
> > Best you can do is collect a full debug[1] log and see when the issue is
> > introduced.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> >
> Do you mind trying the following branch[2]?  Not sure if it will help, 
> but I made some changes to chan_iax2 a few months ago.
> 
> [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

I can try but i have same issue with chan_sip channel also.  and next we have 
scheduled to put this box 1.8.3.2 in production :(  

-S 


> Date: Fri, 8 Apr 2011 13:16:30 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 12:56 PM, Paul Belanger wrote:
> > On 11-04-08 11:55 AM, satish patel wrote:
> >>
> >> @Paul - many time i am gettting following SIP error when channel isn't
> >> available. I want to get rid on this revers thing. I tried all version
> >> 1.8.1,1.8.2,1.8.3 but not fix :(
> >>
> > Best you can do is collect a full debug[1] log and see when the issue is
> > introduced.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> >
> Do you mind trying the following branch[2]?  Not sure if it will help, 
> but I made some changes to chan_iax2 a few months ago.
> 
> [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] IAX2/0.0.29.199

[Paul Belanger]

On 11-04-08 12:56 PM, Paul Belanger wrote:

On 11-04-08 11:55 AM, satish patel wrote:


@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(


Best you can do is collect a full debug[1] log and see when the issue is
introduced.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Do you mind trying the following branch[2]?  Not sure if it will help, 
but I made some changes to chan_iax2 a few months ago.


[2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/

--
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twitter: pabelanger | IRC: pabelanger (Freenode)
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Re: [asterisk-users] IAX2/0.0.29.199

[Paul Belanger]

On 11-04-08 11:55 AM, satish patel wrote:


@Paul - many time i am gettting following SIP error when channel isn't 
available. I want to get rid on this revers thing. I tried all version 
1.8.1,1.8.2,1.8.3 but not fix :(

Best you can do is collect a full debug[1] log and see when the issue is 
introduced.


[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

Look at this sip debug its saying something related Retransmitting #1 (no NAT) 
to 0.0.29.200:5060:

<>
-- Executing [7624@from-sip:1] Macro("SIP/7527-00c2", 
"stdexten,7624,SIP/7624") in new stack
-- Executing [s@macro-stdexten:1] Dial("SIP/7527-00c2", 
"SIP/7624&IAX2/7624,20,t") in new stack
  == Using SIP RTP CoS mark 5
[Apr  8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" ;tag=as6f6822ba
To: 
Contact: 
Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1407056235 1407056235 IN IP4 172.30.1.47
s=Asterisk PBX 1.8.3.2
c=IN IP4 172.30.1.47
t=0 0
m=audio 16720 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Apr  8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
-- Called 7624
-- Called 7624
Retransmitting #1 (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" ;tag=as6f6822ba
To: 
Contact: 
Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257




> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> >  -- Executing [7623@from-sip:1] Macro("SIP/7527-006b", 
> > "stdexten,7623,SIP/7623") in new stack
> >  -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b", 
> > "SIP/7623&IAX2/7623,20,t") in new stack
> >  -- Hungup 'IAX2/0.0.29.199:4569-5255'
> >  -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b", 
> > "IAX2/0.0.29.199:4569-5255") in new stack
> >  -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in 
> > new stack
> >  -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8?  Are you using realtime?  Looks to be an issue with 
> netsock2.c.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

@Paul - many time i am gettting following SIP error when channel isn't 
available. I want to get rid on this revers thing. I tried all version 
1.8.1,1.8.2,1.8.3 but not fix :(


[Apr  8 11:52:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x2f40580 (len 793) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  8 11:52:26] NOTICE[13912]: chan_iax2.c:4643 __auto_congest: 
Auto-congesting call due to slow response

-Satish 

> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> >  -- Executing [7623@from-sip:1] Macro("SIP/7527-006b", 
> > "stdexten,7623,SIP/7623") in new stack
> >  -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b", 
> > "SIP/7623&IAX2/7623,20,t") in new stack
> >  -- Hungup 'IAX2/0.0.29.199:4569-5255'
> >  -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b", 
> > "IAX2/0.0.29.199:4569-5255") in new stack
> >  -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in 
> > new stack
> >  -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8?  Are you using realtime?  Looks to be an issue with 
> netsock2.c.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _
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Re: [asterisk-users] IAX2/0.0.29.199

[satish patel]

No I am not using any realtime config. its text file..

shirley*CLI> core show settings

PBX Core settings
-
  Version: 1.8.3.2
  Build Options:   LOADABLE_MODULES
  Maximum calls:   250 (Current 0)
  Maximum open file handles:   Not set
  Verbosity:   3
  Debug level: 0
  Maximum load average:0.00
  Minimum free memory: 0 MB
  Startup time:15:08:59
  Last reload time:15:08:59
  System:  Linux/2.6.32-24-server built by root on x86_64 
2011-03-22 18:38:19 UTC
  System name:
  Entity ID:   00:30:48:77:1c:3c
  Default language:en
  Language prefix: Enabled
  User name and group: asterisk/asterisk
  Executable includes: Disabled
  Transcode via SLIN:  Enabled
  Internal timing: Enabled
  Transmit silence during rec: Disabled
  Generic PLC: Enabled

* Subsystems
  -
  Manager (AMI):   Enabled
  Web Manager (AMI/HTTP):  Disabled
  Call data records:   Enabled
  Realtime Architecture (ARA): Disabled

* Directories
  -
  Configuration file:
  Configuration directory: /etc/asterisk
  Module directory:/usr/lib/asterisk/modules
  Spool directory: /var/spool/asterisk
  Log directory:   /var/log/asterisk
  Run/Sockets directory:   /var/run/asterisk
  PID file:/var/run/asterisk/asterisk.pid
  VarLib directory:/var/lib/asterisk
  Data directory:  /var/lib/asterisk
  ASTDB:   /var/lib/asterisk/astdb
  IAX2 Keys directory: /var/lib/asterisk/keys
  AGI Scripts directory:   /var/lib/asterisk/agi-bin




> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> 
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> >  -- Executing [7623@from-sip:1] Macro("SIP/7527-006b", 
> > "stdexten,7623,SIP/7623") in new stack
> >  -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b", 
> > "SIP/7623&IAX2/7623,20,t") in new stack
> >  -- Hungup 'IAX2/0.0.29.199:4569-5255'
> >  -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b", 
> > "IAX2/0.0.29.199:4569-5255") in new stack
> >  -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in 
> > new stack
> >  -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8?  Are you using realtime?  Looks to be an issue with 
> netsock2.c.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
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Re: [asterisk-users] IAX2/0.0.29.199

[Paul Belanger]

On 11-04-08 10:48 AM, satish patel wrote:


Where this revers IP comes from ?

   == Using SIP RTP CoS mark 5
 -- Executing [7623@from-sip:1] Macro("SIP/7527-006b", 
"stdexten,7623,SIP/7623") in new stack
 -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b", 
"SIP/7623&IAX2/7623,20,t") in new stack
 -- Hungup 'IAX2/0.0.29.199:4569-5255'
 -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b", 
"IAX2/0.0.29.199:4569-5255") in new stack
 -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in new 
stack
 -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'

Asterisk 1.8?  Are you using realtime?  Looks to be an issue with 
netsock2.c.


--
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Re: [asterisk-users] iax2 sound problem

[Oğuzhan Kayhan]

Hello,
An update
I recompiled all asterisk from the beginning.
I had similar issues, then I noticed something.
I use zoiper kiax and diax as clients.
In kiax and diax I got no audio in any iax calls.
But in zoiper, When I "register" the client, no audio.
But if I unregister the client, I can dial and call anywhere with iax.

This is registered call via zoiper (no audio)

Registered IAX2 '6001' (AUTHENTICATED) at 178.233.186.16:4569
-- Accepting AUTHENTICATED call from 178.233.186.16:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (ulaw|alaw|gsm),
   > priority = mine
-- Executing [*66@default:1] Playback("IAX2/6001-2657", "demo-echotest")
in new stack
--  Playing 'demo-echotest.ulaw' (language 'en')
  == Spawn extension (default, *66, 1) exited non-zero on 'IAX2/6001-2657'
-- Hungup 'IAX2/6001-2657'

and this is the call after unregistreing zoiper. I got nice auido with
that..

Unregistered IAX2 '6001' (AUTHENTICATED)
-- Accepting AUTHENTICATED call from 178.233.186.16:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (ulaw|alaw|gsm),
   > priority = mine
-- Executing [*66@default:1] Playback("IAX2/6001-1243", "demo-echotest")
in new stack
--  Playing 'demo-echotest.ulaw' (language 'en')
  == Spawn extension (default, *66, 1) exited non-zero on 'IAX2/6001-1243'
-- Hungup 'IAX2/6001-1243'

no difference in verbose mode.
Any way I can try to find the problem??



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
> Sent: Monday, March 21, 2011 11:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] iax2 sound problem
> 
> Hello,
> I installed 1.6.2.17 version of asterisk.
> Set the user database to realtime.
> 
> I have no problems with sip users.
> They can register talk etc..
> With iax clients, they can register also.. And when they call iax to
> sip, it
> works.
> When they make an echo test..no voice received on iax clients.
> When they make call from sip to iax ..no sound received on iax clients.
> 
> I didnt see any clue on debug.
> I tried different clients as zoiper kiax but no chance..
> As codec ulaw alaw gsm..no luck.
> 
> 
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Re: [asterisk-users] IAX2 and INVAL packets

[Sebastian]

On 11/18/2010 08:01 PM, Sebastian wrote:
> Is anybody here familiar with the meaning of INVAL packets for IAX2?
>
> Every few days I get a dropped outgoing call in the middle of the
> conversation (the outgoing call has been connected for few minutes) when
> an incoming call comes in. The log reads the following when this happens:
>

Just answering myself here. After spending few months troubleshooting 
this completely erratic fault - it turns out that it was the ADSL 
router. I still don't know for certain the exact mechanism of the fault 
- but it looks like my IAX provider would receive some packets which 
were invalid from my direction - and hang-up calls randomly. There was 
no rhyme or reason for it. Sometime it would happen few times a day, 
sometime once every few days. Sometime 10 seconds into the call, 
sometime when another call would come in. But it always seemed to be 
connected with the "received INVAL" message in the logs.

I replaced the router (a TP-Link ADSL router) with a Belkin router 
(that's what I had to hand) and everything is fine now for more then a 
week. The other router did not give any signs of trouble otherwise - all 
other Internet, email, openvpn and any other traffic going through it 
was fine. But somehow it would generate this problem with Asterisk.

I just thought it might help somebody some day.

Sebastian

>
>
> [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963,
> having received INVAL
> [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963
> [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up
> IAX2/ihs_trunk_out-2963 now...
> [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup
> 'IAX2/ihs_trunk_out-2963'
>
>
>
> And more setup details, for those who still have the will to live :-)
>
> Asterisk version: 1.6.2.13
> Internal externsions: everything on SIP - 3 Grandstream GXP-2000, 2
> analog phones on a pci OpenVox card and 2 Linphone softphones
> Trunks: IAX2
> Trunks provider: Gradwell
> Asterisk machine: 800Mhz Intel Pentium, 512MB of RAM
> Internet connection: Tiscali business ADSL
>
> I am happy to post here any config files and logs you might think would
> be relevant.
>
> This is not consistent - and I've managed to have 4 concurrent calls
> which held 30 minutes (before I hung them up) when I tried. So not easy
> to replicate.
>
> Sebastian
>

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Re: [asterisk-users] IAX2 and INVAL packets

[Sebastian]

On 11/18/2010 10:38 PM, Tilghman Lesher wrote:
> On Thursday 18 November 2010 14:01:49 Sebastian wrote:
>> >  Is anybody here familiar with the meaning of INVAL packets for IAX2?
>> >
>> >  Every few days I get a dropped outgoing call in the middle of the
>> >  conversation (the outgoing call has been connected for few minutes) when
>> >  an incoming call comes in. The log reads the following when this
>> >  happens:
>> >
>> >
>> >
>> >  [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963,
>> >  having received INVAL
>> >  [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963
>> >  [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up
>> >  IAX2/ihs_trunk_out-2963 now...
>> >  [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup
>> >  'IAX2/ihs_trunk_out-2963'
>> >
>> >
>> >
>> >  And more setup details, for those who still have the will to live:-)
>> >
>> >  Asterisk version: 1.6.2.13
>> >  Internal externsions: everything on SIP - 3 Grandstream GXP-2000, 2
>> >  analog phones on a pci OpenVox card and 2 Linphone softphones
>> >  Trunks: IAX2
>> >  Trunks provider: Gradwell
>> >  Asterisk machine: 800Mhz Intel Pentium, 512MB of RAM
>> >  Internet connection: Tiscali business ADSL
>> >
>> >  I am happy to post here any config files and logs you might think would
>> >  be relevant.
>> >
>> >  This is not consistent - and I've managed to have 4 concurrent calls
>> >  which held 30 minutes (before I hung them up) when I tried. So not easy
>> >  to replicate.
> An INVAL response basically means that the remote Asterisk box received
> a packet for a call that it did not think existed.  So likely, something
> else caused the call to hangup (such as an unrelated error crashing a
> process, and the replacement process had no record of such a call, so it
> sent an INVAL response to any subsequent packet).
>
> Technically, this could also be done as a MITM attack.  If something were
> to see even a single packet related to the call, it is able to fake an
> INVAL packet.  BTW, this is not unique to IAX2; a MITM attack can also fake
> a SIP CANCEL.

Hi Tilghman and thank you for replying. I have been working on narrowing 
this down for a few months now - without much success.

Do you have any suggestions on taking further steps to find the cause?

In case it helps:

1. I use iptables to only allow in IAX2 connections from the IP 
addresses of the VoIP provider (Gradwell).
2. I also restrict incoming connections in iax.conf only to the same ip 
ranges.

The drops occur randomly, once every few days normally (but there have 
been some cases of few drops in one day).

Again, not sure if it is relevant - but here is the same log - only 
including few extra earlier lines. The strange thing is that, just 
before hanging up the call I'm interested in (2963), it seems to be 
hanging up another call - it reads "destroying 706" - but I can't find 
any reference to a call "706" anywhere earlier in the log. So I can't 
understand what call is that, and why does it get hung-up:


[Nov 17 15:25:01] DEBUG[5137] chan_iax2.c: Determining if address 
212.11.91.202 with username x_in requires calltoken validation. 
  Optional = 1  calltoken_required = 0
[Nov 17 15:25:01] DEBUG[5137] chan_iax2.c: ip callno count incremented 
to 1 for 212.11.91.202
[Nov 17 15:25:01] DEBUG[5132] chan_iax2.c: Immediately destroying 706, 
having received hangup
[Nov 17 15:25:01] DEBUG[5132] chan_iax2.c: schedule decrement of callno 
used for 212.11.91.202 in 60 seconds
[Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, 
having received INVAL
[Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963
[Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up 
IAX2/ihs_trunk_out-2963 now...
[Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup 
'IAX2/ihs_trunk_out-2963'



Any suggestion to help take this further is much appreciated.

Sebastian

>
> -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter:
> Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at:
> www.digium.com & www.asterisk.org
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Re: [asterisk-users] IAX2 works one direction, but not the other...

[bakko]

Hi Paul,

maybe there is some think wrong on iax.

if I set remotesecret on IAX2 extension the call from Server A to Server B 
work but not authenticated and host is set to dynamic (normaly if is a IP 
authentication on host parametre I put the IP)

If I set secret on two box and both are registered without errors, when i 
call from box A or from box B to box A, always I receive this error "No 
auhthority found".

Thi behaviour only happens if the Asterisk version onTwo box is 1.6.2.13. If 
Asterisk version on Box A is 1.4.X and Asterisk version on Box B is 1.6.2.13 
(with the same configuration) work fine.

Why?

Thank you in advance.

Regards

- Bakko


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Re: [asterisk-users] IAX2 works one direction, but not the other...

[khalid touati]

Hi Cassius,
it may be slightly offsubject but i did connect two 1.6 asterisk boxes, and
the only issue i had is these two statements missing:
calltokenoptional=209.16.236.73/255.255.255.0
requirecalltoken=no

hope it helps!

On Tue, Oct 19, 2010 at 12:54 PM, Paul Belanger <
paul.belan...@polybeacon.com> wrote:

> On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith 
> wrote:
> > Any hints for me?
>
> Server Ottawa (192.168.1.190)
>
> register => 
> Ottawa:ottawaisc...@192.168.1.196
>
> [Toronto]
> type=peer
> host=dynamic
> username=Toronto
> secret=TorontoIsFine
>
> Server Toronto (192.168.1.196)
>
> register => 
> Toronto:torontoisf...@192.168.1.190
>
> [Ottawa]
> type=peer
> host=dynamic
> username=Ottawa
> secret=OttawaIsCool
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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>
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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Paul Belanger]

On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith  wrote:
> Any hints for me?

Server Ottawa (192.168.1.190)

register => Ottawa:ottawaisc...@192.168.1.196

[Toronto]
type=peer
host=dynamic
username=Toronto
secret=TorontoIsFine

Server Toronto (192.168.1.196)

register => Toronto:torontoisf...@192.168.1.190

[Ottawa]
type=peer
host=dynamic
username=Ottawa
secret=OttawaIsCool

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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Paul Belanger]

> I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with
> username and password.
>
> Only when i changed secret with remotesecret the connection work.
>
I would enable iax debugs and confirm you calls you being
authenticated, and not using a guest account.  As I mentioned,
'remotesecret' is not an option for chan_iax2.so

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Re: [asterisk-users] IAX2 works one direction, but not the other...

[bakko]

Hi Paul,

I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with 
username and password.

Only when i changed secret with remotesecret the connection work.

Maybe you can try the same configuration to confirm this behaviour

Regards

- Bakko 


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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Paul Belanger]

On Tue, Oct 19, 2010 at 10:44 AM, Andrea Sannucci  wrote:
> On the iax conf for both box change the secret parameter with remotesecret.
> This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X
>
This is incorrect, chan_iax.so does not have such a parameter.
However, chan_sip.so does, and it is documented.

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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Andrea Sannucci]

Hi Cassius

On the iax conf for both box change the secret parameter with remotesecret.

This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X

Regards

2010/10/18 Cassius Smith 

> I'm having trouble getting an IAX2 connection between a couple of
> servers. I
> can make calls from server B to server A, but when I call from Server A
> to server
> B, I get "No authority found".
>
> If I remove serverA's password on ServerB's iax.conf, calls will go
> through as "UNAUTHENTICATED".
>
> On ServerA I am running Asterisk 1.6.2.9
> On ServerB I'm running 1.6.2.13
>
> Any hints for me?
> The registrations in both directions seem to work fine when I do an iax2
> reload from the CLI.
>
> config file snips shown below.
> Thanks
> Cassius Smith
> =
>
> On server B, I have the following:
> [general]
> register => serverB:longsecretpasswo...@servera_ip
>
> [serverA]
> type=friend
> host=dynamic
> auth=md5
> secret=longsecretpassword1
> context=no911
>
> [serverB]
> type=friend
> host=dynamic
> auth=md5
> secret=longsecretpassword2 ; if I remove this, calls go through as
> UNAUTHENTICATED
> context=no911
>
> On server A, I have the following:
> [general]
> register => serverA:longsecretpasswo...@serverb_ip
>
> [serverB]
> type=friend
> host=dynamic
> auth=md5
> secret=longsecretpassword2
> context=no911
>
> [cary]
> type=friend
> host=dynamic
> auth=md5
> secret=longsecretpassword1
> context=no911
>
>
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Re: [asterisk-users] IAX2 works one direction, but not the other...

[Cassius Smith]

BTW I apologize for the double send. 




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