Re: [asterisk-users] To Header instead of Request URI based routing

[Max Grobecker]

Hi,

do you have access to the system that sends you these calls?
If it's also an Asterisk, you could tell it to send another INVITE URI, 
regardless of what is submitted
in the registration.

On Asterisk with chan_sip you can do it by dialling:

 Dial(SIP/your_peer/+49202thatgoesinthetouri!+49202thatgoesintheinviteuri)

That is, as said, if the remote system which is sending you the calls is an 
Asterisk machine so you can
just reconfigure the way you get the calls to your local machine.

If it's not your system, you need to parse the To: header - for example, with:

 Set(ToHeaderVal=${SIP_HEADER(To)})
 Set(DailedNumber=${CUT(ToHeaderVal,:,2)})
 Set(DailedNumber=${CUT(DailedNumber,@,1)})

That should give you the dialed number in Variable "DialedNumber".


Greetings
 Max




Am 22.12.2017 um 14:54 schrieb Benoit Panizzon:
> Dear List
> 
> It looks like the common way to to sip signaling over a trunk is:
> 
> In the Request URI, return the 'Register' Contact.
> In the To: Header, send the destination number.
> 
> Unfortunately, asterisk with pjsip (i did not try chan_sip) does
> expect the dialed extension as request uri and does ignore what it is
> getting in the To: header.
> 
> I could not find any hint in the documentation of this can be changed.
> 
> I found instructions for a work-around:
> 
> http://www.kempgen.net/voip/sip-request-uri-vs-to-header-routing.html
> 
> In the meantime: Is there a way to tell the asterisk with pjsip to use
> the To: header to address an extension?
> 
> Kind regards
> 
> -Benoît Panizzon-
> 



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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Max Grobecker]

Hi Luca,


Am 06.05.2017 um 15:49 schrieb Luca Bertoncello:
 
> I'm running an own BIND on my Linux-PC...

Me too ;-)


> Maybe should I configure a forwarder for the zone t-online.de? It not
> difficult, and if you mean it can help, I'll do that...

In the meantime, I setup forwarding requests to "t-online.de" and "t-ipnet.de" 
to the address 194.25.2.129.
That is kind of a global DNS resolver for all customers and is working since 
the 90s without address changes.



> Could you say me how can I disable the SRV lookups?
> I use Asterisk 1.8.30.0 on an OpenWRT device.

In your sip.conf, simply add

srvlookup = no

To your DTAG peer configuration.
If set globally, you may break the ability to directly call SIP addresses.



> The version of Asterisk on my OpenWRT unfortunately does not support dnsmgr...


On embedded systems, I often had problems with "stuck" DNS.
But that was ages ago... The last time on my old "Horstbox" with Asterisk 1.2 
and bristuff on Linux 2.4 :-/

Have you rebooted the whole WRT device or just restarted the Asterisk service 
to resolve your problem?
Maybe it's less an Asterisk issue but one with DNS caching on this device?



Viele Grüße aus dem Tal
 Max



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Re: [asterisk-users] Need to restart Asterisk if remote server not working?

[Max Grobecker]

Hello,

I'm also a customer of the DTAG.
Yesterday, the messed a bit with their DNS entries...

If you are NOT using their DNS resolvers you got a "wrong" IP address back that 
was not working.
Besides that, you should disable SRV lookups for their SIP peers. Since 
Asterisk's chan_sip.c does not honour the
weight of the SRV entries, nor it failovers to the other records, you might 
just end up with a not working server.
PJSIP might work with that, but it depends on your version.

The "blank" A record for "tel.t-online.de" is also provided and will be changed 
in case of service disruptions on one server,
so it's acceptable to rely on that.

DTAG is providing the following SIP servers at the moment (and also yesterday) 
with their SRV records:

_sip._udp.tel.t-online.de. 401  IN  SRV 0 5 5060 ims001.voip.t-ipnet.de.
_sip._udp.tel.t-online.de. 401  IN  SRV 1 5 5060 ims002.voip.t-ipnet.de.


ims001 should be the preferred one based on the SRV weight. But Asterisk only 
looks at the first record that comes as an answer,
so if ims002 is at the first position it will be used for registration, 
regardless that the other record is weighted better.
And if that one is not answering...
So: Better disable SRV lookups if you are not sure if your SIP channel driver 
supports it ;-)


You should also use the dnsmgr of Asterisk, resp. configuring it to reasonable 
values.
In dnsmgr.conf I set:

enable = yes
refreshinterval = 10


If dnsmgr is not enabled on your server this might have caused the problem 
because your SIP driver did not recognized that the
target address of the configured hosts has changed.
DNS changes should work also without dnsmgr - but since I've enabled the dnsmgr 
I had far less problems with changing DNS records ;-)



Am 06.05.2017 um 09:37 schrieb Luca Bertoncello:
> Hi list!
> 
> Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't
> connect to the remote Server (by Telekom) until today about 7:30.
> 
> Well, it could happen...
> What I find really annoying was that I needed to restart Asterisk as I
> checked with sipsak that the Telekom-Server works...
> 
> I think, this should not be normal... Can someone explain me why it happens
> and what I have to change in the configuration to avoid this problem?
> 
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
> 



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Re: [asterisk-users] How to have callers not being billed when in waiting queue ?

[Max Grobecker]

Hi,

in Germany, this kind of regulation is in effect for phone numbers which cost 
more than a normal landline call.
The regulation states, that the waiting time must not be charged to the 
customer.


Most companies implemented this by simply switching their telephone numbers to 
those, which are charged per call 
(so there's no difference in price between waiting for someone to pick up or 
being connected to someone) ;-)
Or they decided to use a normal landline phone number for which this regulation 
does not apply.

The second method was to not answer the call before really connected to a 
person on the queue and using Early Media as you mentioned.
But: The maximum length of this Early Media stream is in most telephone 
networks limited to somewhat around 90 to 180 seconds,
then the call gets disconnected by the network.

I'm not very familiar with regulations and numbering plans in France, but maybe 
there's also something called "offline billing".
Using this, your call is not billed by the caller's telephone company until you 
send them the amount of time that should be billed for a specific call.


Your best choice will be, that - if you ever get those regulations - you should 
rely on what your telephone number provider tells you to do ;-)


Greetings
 Max


Am 28.03.2017 um 15:24 schrieb Olivier:
> Hello,
> 
> In France, years ago, there was some discussions about a new regulation 
> forcing some providers to not charge anything to callers while those are 
> waiting for a call center agent to become available.
> Once caller and agent are on call with each other, nominal charging applies.
> 
> No matter if those discussions ever did or didn't change current regulation, 
> I wonder which dialplan statements could technically comply this dual billing 
> requirement ?
> 
> 
> same = n,Progress()
> same = n,Queue(whatever,...,macro-option, ...)
> 
> To me, coupling Progress app with Queue's  macro or gosub option like above, 
> would let a sysadmin answer a queued call.
> Doing so, time spent before connection with queue agent should not be billed 
> to anyone (caller nor callee), while time spent after connection is billed 
> normaly.
> 
> 1. Should this work ? Am I missing something ?
> 
> 2. Is there an alternative way to implement this ?
> 
> 3. Comments ? Suggestions ?
> 
> Regards
> 
> 



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Re: [asterisk-users] multiple outbound invites

[Max Grobecker]

Hi,

that could be caused when your upstream offers "100rel" and your Asterisk does 
not get a response fast enough from your upstream.
Is your outbound peer monitored by the qualify feature (qualify=yes)? 
Then asterisk should calculate the round-trip-time until a response arrives and 
should not resend the packets too fast.
If that does not help, you could play around with the "timert1" settings in 
your peer's SIP configuration.

Also, this sounds to me like a bug on the carriers side. It seems they are 
maybe offering "100rel" to you, but do not send any SIP/1xx responses
regarding your INVITE so your Asterisk resends these INVITEs because they are 
assumed lost.
Since these INVITEs all have the very same Call-ID and CSeq number, your 
carrier's equipment should be able to determine these packets are regarding
the same call. Again, I think your carrier should fix this problem on his site.
If he wants to enforce rate-limiting to INVITEs he should do it right by 
honouring the Call-IDs and sequence numbers.

If you like you can anyway send me your trace off-list, maybe there's something 
other weird going on.


Greetings
 Max



Am 22.02.2017 um 18:57 schrieb Jeff LaCoursiere:
> 
> Hello,
> 
> I have two upstream providers we use for US termination.  The dialplan sends 
> calls out the "primary" and if that fails for specific reasons, it sends the 
> same call out the "secondary". This has worked well for us when we are lazy 
> about keeping balances up, for example.
> 
> Starting a few days ago ALL calls sent to the 'primary' were returned as 
> busy, though the secondary terminated them fine.  We have a balance, and 
> funny enough international calls are going through fine, just not US calls.  
> I opened a ticket.
> 
> The response form the carrier is that our asterisk is sending four 
> simultaneous invites within one second, and for that reason the call is 
> rejected.
> 
> I did a packet trace and was able to confirm this is true - only US calls 
> sent to this carrier cause our end to send four identical simultaneous 
> invites.  When it fails, a single invite for the same call is sent to the 
> secondary, which is terminated without issue.
> 
> Happy to send the SIP trace if any would care to see it, but is there a 
> reason anyone can think of that our asterisk (11.11.0) would suddenly start 
> doing this?  It may be that it has been doing it all along, and our carrier 
> just started rejected calls that come in this way, I'm not sure.
> 
> Cheers,
> 
> j
> 
> 

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Am 16.02.2017 um 15:01 schrieb Joshua Colp:

> As for your issues please do file them. I'd also suggest using bundled
> PJSIP, it works the best with Asterisk and we backport applicable fixes
> and include fixes we've created that have not yet made it to a PJSIP
> release.

OK, I'll try again with the bundled version.
If the bugs persist, I'll file some bugs ;-)


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hi,

Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> And Microsip using PJSIP SIP stack :)

Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not 
sure if this really is a sign of good quality.
Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP 
in other software), but after just five minutes of testing
I found several bugs regarding PJSIP preventing me to use it in a production 
enviroment :-(
I'm going to file these bugs at the moment...


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:
> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
> security reasons I have enable TLS on our Asterisk server to support TLS 
> authentication, It works well with hard phones. Has anybody in this forum use 
> SIP Soft phones with TLS authentication enabled? Any suggestions?
> 
>  
> 
> Thanks,
> Motty
> 
> 
> 



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Re: [asterisk-users] SIP host name resolution

[Max Grobecker]

Hi,


Am 03.02.2017 um 18:23 schrieb Steve Edwards:
> If I have a SIP endpoint defined in sip.conf using a host name instead of an 
> IP address, do I have to reload sip to get Asterisk to 're-resolve' the host 
> name if I change the IP address in my DNS?

Normally, Asterisk honours DNS TTL and will re-lookup hosts as soon as the TTL 
is expired.
If you can't wait for that to happen, you can enable the builtin DNS manager 
and configure a refresh interval for DNS records to expire.
See "dnsmgr.conf" for the latter one.


> Does the answer change if the host name in sip.conf resolves to a CNAME and I 
> change the CNAME in my DNS?

Not as far as I know.
If you enabled SRV lookups for Asterisk, you may also want to check possibly 
existing SRV records for your host since Asterisk then looks them up first.


Max

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Re: [asterisk-users] Connection dropped after 15 minutes with Deutsche Telekom

[Max Grobecker]

Hi,

I figured out that this happens, when Asterisk ignores Session-Timers requests.
So I added the following to my DTAG peer configuration and eleminated the 
problem - and can use g722 on the DTAG network :-)

---
session-timers=accept
session-expires=120
auth_options_requests=yes
---


Greetings
 Max


Am 08.01.2017 um 19:47 schrieb Luca Bertoncello:
> Luca Bertoncello <lucab...@lucabert.de> schrieb:
> 
> Hi again!
> 
>> The problem: after 15 minutes will the call dropped, but only if the call is
>> to another nation! If I just call another phone in Germany, I can speak
>> longer than 15 minutes...
> 
> After a long work, and with the huge help of Michael Maier, I found the
> problem...
> I write here the description of the problem and my solution, maybe can this
> help someone other having the same problem...
> 
> The problem: after a successfully INVITE with the complete list of all
> supported Codecs, I receive about 15 minutes after call start, another INVITE
> (re-INVITE) from Telekom with __JUST__ one Codec: the one used by the call
> (currently: alaw).
> My Asterisk sends an "200 OK" with the same Codec and Telekom apparently has
> a problem with my answer, since the connection will be closed...
> 
> __MY__ solution: I configured Asterisk to use just __ONE__ Codec (alaw) for
> the communication with Deutsche Telekom.
> Now it seems to work, then I can call Italy and can speak longer than 15
> minutes.
> 
> I'm really puzzled and can't understand why Telekom has problem with my
> answer __JUST__ on calls outside Germany, but that is...
> 
> So, if someone other has the same problem, can try with my solution.
> 
> Hope to help!
> 
> Regards
> Luca Bertoncello
> (lucab...@lucabert.de)
> 

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Re: [asterisk-users] new inbound DID provider... no auth?

[Max Grobecker]

Hi,

That's right - you just need to define a peer with a static IP address and 
"type=peer" to assign incoming calls to a peer name and apply
the corresponding configuration (e.g. codecs).
To make your configuration less redundant you can use templates in your peer 
definition
(at least for chan_sip, I'm not sure if the same syntax applies on chan_pjsip).

Example:

---

;; All configuration made to this peer will be applied to all childs of this 
definition
[your-did-provider](!)
type=peer
allow=ulaw,alaw,g722
...

;; This peer derives all other configuration from "your-did-provider",
;; then your local changes are applied and can override the derived ones.
[your-did-provider-gw1](your-did-provider)
host=1.2.3.4

[your-did-provider-gw2](your-did-provider)
host=1.2.3.5
---

That's the shortest thing I can imagine at the moment.
At least, with this way of definition you only need to do changes on one single 
point, not for every gateway IP.





Am 30.11.2016 um 22:10 schrieb Jeff LaCoursiere:
> 
> We are trying to work with a new DID provider and I find myself confused.  
> Their standard integration is to send the call with no authentication.  I am 
> expected to whitelist all their possible gateways, and accept their calls I 
> guess with no peer definition.  I actually have it working this way; the 
> calls land in our "public" context, I guess as "guest", and I am able to 
> route them from there.  But that makes me nervous.
> 
> I would rather at least have them be associated with a defined peer, so I can 
> set the right context and any other parameters I might want associated.  It 
> is inbound only, no outbound.  I might try to set a host= in a peer 
> definition with no secret, and see if that matches it, but I would rather 
> avoid making a peer definition for every gateway they have.  Can anyone think 
> of a way to define a single peer that might show from multiple potential 
> addresses without authentication info?
> 
> Cheers,
> 



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Re: [asterisk-users] Change Media IP in SDP

[Max Grobecker]

Hi,

normally, Asterisk handles RTP IP addresses in SDP correctly, if you have 
specified
 - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia)
 - your local network with "localnet=yournetwork/networkmask" - e.g. 
"localnet=192.168.1.0/255.255.255.0"
 - directmedia, canreinvite, directrtpsetup is deacitivated

In this case, your Asterisk will always stay in the RTP stream and signalling 
only it's own IP address to other peers
which is configured for the interface which Asterisk uses to reach the peer.


But: If you have only one interface configured on your Asterisk server and an 
external firewall/router is managing your separated networks, this might not 
help.
In this case you can use "externip" on a per peer basis in your SIP 
configuration to specify the IP address Asterisk uses in the SDP.
Maybe, a global configuration of "externip" and "localnet" is all you need to 
help Asterisk setting the SDP address correctly.
Also, enabling ICE support can help you getting the correct IP address if the 
remote peer supports it.



Greetings
 Max


Am 07.12.2016 um 00:02 schrieb Harel:
> Hello List,
> I need your help with information going out on my SDP.
> Is it possible to update the Media Address on a per-call basis or a 
> per-channel basis?
> Reason:
> My Asterisk is in a private network and needs to connect to UA on its 
> internal network and also few external networks. One network is public and 
> the others are not public. Between each other the external networks are not 
> routable. Signaling is flowing with no issues because SIP Registers and NAT 
> boxes maintain sessions correctly. The problem is with RTP. After making 
> traces on all possible nodes of this network I clearly found out that the RTP 
> fails because the Asterisk doesn't manage to communicate the correct address 
> to the UAs in the SDP. It will report its internal IP address and the remote 
> UA will try to send its RTP to this address which, of course, will fail 
> miserably. 
> Obviously I can't use externaddr or media_address in sip.conf because it will 
> only be good for one network while the other external networks will fail just 
> the same. Same applies for STUN, it will only be good for the network the 
> STUN requests are being sent from. 
> On all networks I have fix IP addresses on my side and I fully control a 
> professional security box. 
> Asterisk is 13.6.0
> I can't, and don't want to, touch user-side equipment which is normally some 
> kind of voip phone behind a standard home VDSL router.
> 
> Any ideas how can I transmit the correct IP address in SDP to UAs on 
> different networks?
> 
> Many thanks,
> Harel
> 
> 



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Re: [asterisk-users] Touch tone stutter

[Max Grobecker]

Hi,

you could try switching the DTMF mode of the ATA's SIP peer (and also in the 
ATA itself) to INBAND transmission.
In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk 
can interpret it.
For this to work, the ATA needs to use a G.711 codec. Inband DTMF needs an 
uncompressed codec to work properly.

Another way is (if the ATA supports it) to switch DMTF mode to SIP INFO.
In this mode, DTMF is not interpreted out of the audio stream. For external 
peers which are not supporting this mode
Asterisk then generates the proper RTP messages or tones.

With SIP INFO mode I made my best results with all devices, sadly it's not very 
common used.


Max


Am 23.11.2016 um 20:02 schrieb D'Arcy Cain:
> On 2016-11-22 07:49 PM, Pete Mundy wrote:
>>
>> One direction that may be worth exploring further is his ATA's config (or 
>> perhaps swapping it for a different model). Eg adjusting echo cancellation 
>> or line impedance settings.
> 
> I have to be careful here as I auto-provison these devices and changes would 
> propogate to every user.  Echo cancellation is off.  Do you think it should 
> be on?
> 
>> Is the ATA he is using the same as the ATA you use?
> 
> No but it is the same as other users who do not have the problem.  I use a 
> SIP phone and a Cisco ATA.
> 
>> Failure to correctly recognise and decode DTMF is just one of many reasons 
>> why I never use them (ATAs). Like faxing over VoIP, they're just too much 
>> trouble :(
> 
> I understand but some use cases just need it.
> 
>> Genuine IP phones are pretty good value these days. Could you drop one of 
>> those on-site as a temporary measure to prove that it's phone and/or ATA 
>> related?
> 
> He does want to have an extension so that won't work.
> 
>> Ps, you might also want to consider joining VoiceOps (if you're not already 
>> subscribed) and posting there. 
>> https://puck.nether.net/mailman/listinfo/voiceops
> 
> I have subscribed.  Thanks.
> 



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Re: [asterisk-users] Non-global variable that follows channel?

[Max Grobecker]

Hi,

is channel variable inheritance working for your setup?
Passing variables to other channels can normally simply be done by naming the 
variable with one or two prefixed undersorces
to make it available to the channel that is created from that one defining the 
variable.
But I have no idea if it's getting inherited to Gosub called from a Dial 
command...
 -> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Set

If that is not working for you, you might use the SHARED() variables which are 
kind of global accessible by the channel ID.
So you might call your Gosub with only the (unique) reference name of the 
variables you wish to pass and then call it from your Gosub.
 -> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHARED


Greetings,
 Max


Am 23.11.2016 um 13:06 schrieb Jonathan H:
> Related to 
> http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html,
> at the moment I'm passing one variable via DIAL.
> 
> Now I'd like to pass a whole bunch, and my idea was to rather than
> having a great string of
> 
> b(synctest3b^setVar^1(something)^2(more things)^3(etc))
> 
> and then get them with ARG1..ARGn etc, I could bundle the whole lot
> into a HASH and then unbundle them at the called channel.
> 
> Passing the HASH as a var isn't working (I wasn't expecting it to!)
> but is there any other way of doing this, or is it setVar for each
> one?
> 



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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Max Grobecker]

Hi,

Am 17.11.2016 um 13:51 schrieb Jerry Geis:

> PBX Core settings
> -
>   Version: 11.24.1
>   Build Options:   LOADABLE_MODULES, BUILD_NATIVE
>   Maximum calls:   Not set
>   Maximum open file handles:   1024
>   Root console verbosity:  0
>   Current console verbosity:   5
>   Debug level: 0
>   Maximum load average:0.00
>   Minimum free memory: 0 MB
>   Startup time:16:23:00
>   Last reload time:16:23:00
>   System:  Linux/2.6.32-642.6.2.el6.x86_64 built by root 
> on x86_64 2016-10-30 20:40:02 UTC
>   System name:
>   Entity ID:   b0:83:fe:d1:af:5d
>   Default language:en
>   Language prefix: Enabled
>   User name and group: /
>   Executable includes: Disabled
>   Transcode via SLIN:  Enabled
>   Transmit silence during rec: Disabled
>   Generic PLC: Enabled
>   Min DTMF duration::  80



That's a bit odd... On my Asterisk 11 setup, I see an entry "Internal timing" 
which is totally missing on your installation.
You might want to try adding

  internal_timing = yes

to the [general] section of your asterisk.conf and then stop and start your 
Asterisk.

You can also try to unload all timing modules but "res_timing_timerfd.so" and 
try if it makes things better.
If it does, you can prevent res_timing_dahdi from being loaded in your 
modules.conf.


Max



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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Max Grobecker]

Hi,


Am 15.11.2016 um 17:52 schrieb Olivier:
> Hi,
> 
> How can I double check which timer is currently is use in a running system ?
> core show settings doesn't tell anything, if I'm not mistaken.

To determine which timing module is currently in use, you can take a look at 
"module show like timing".
There should be only one module with "use count" 1 - that's the one that is 
currently used.
If there is no call running, you can unload any additional timing module you 
don't want to use to force Asterisk 
to use the only one left by simply doing "module unload res_".

Also, please check in "core show settings" if internal timing is enabled or 
not. If it's not, please enable it in asterisk.conf.
The internal timing should be enabled by default, but if it's not Asterisk 
might not use any timing module at all if RTP is being bridged between two ends 
of a call.
Asterisk normally synchronises the RTP clocking to one end of the call. But if 
this RTP source is not realiable (jitter, packet loss, silence suppression...) 
you
can end up having audio problems.



Max



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Re: [asterisk-users] SIP and RTP port and IP addresses

[Max Grobecker]

Hi Ethy,


Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:

> How are these parameters available from dialplan?
> 
> For instance, ${SIPURI} holds the internal "IP:port" if the client is behind 
> NAT. 
> I need the external IP:port


You can get the peer's signalling IP address from ${CHANNEL(recvip)} and the 
RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}.
If you need more information (like the codecs used) you can find other channel 
variables on 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL

Please note that, if you have not disabled re-invites, the RTP address may 
change while the call is running.



Max



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Re: [asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

[Max Grobecker]

Hi Jonathan,

Am 05.11.2016 um 14:08 schrieb Jonathan H:

> What I don't understand is that while Ubuntu has IPv6 of course, the VPS host 
> is set to V6 disabled. and as far as I am aware, and my ITSP doesn't have 
> IPv6, so I just can't figure out why two IPv4 systems are getting IPv6 
> "pollution" as it were. And why now??!

That *MAY* be caused by a rogue IPv6 Routing Advertisement in the network where 
your vServer is located.
If you have a global IPv6 address assigned to your interface with the flag 
"dynamic" you got this address via autonomous addressing provided by routing 
advertisement.
To verify, look at the output of:

  sudo ip -6 addr show dev 

You'll find one or more lines starting with "inet6", followed by the assigned 
address and at the end of the line the flags;
For example "inet6 2003:..:1234/64 scope global dynamic" - this would be a 
dynamically assigned address.

Also, doing a

  sudo ip -6 route show default

Will bring more clarity, if you get a route entry like this:
"default via fe80::230:88ff:fe04:d dev ppp0  proto kernel  metric 1024  expires 
1539sec hoplimit 64"

The "expires" information indicates this route has been learned by RA. If you 
have no route entry this means you might have no IPv6 connectivity at all.
If there is a route entry but without "expires" information the route has been 
added manually.


If you have a global IPv6 address assigned to your interface, please check if 
it belongs to your providers network.
An easy way to check this is via https://stat.ripe.net (they use all RIR 
databases, so you'll find information about all regions).


In either way: Your provider should be worried about this.
Either there is a way for other customers to advertise (malicious) IPv6 routing 
information into the network that affects other customers
or your provider simply does not know that he is actively announcing and 
routing IPv6 or configuring customer's vServers with IPv6.

If it's a malicious or at least unknown advertisement, you definitely should 
deactivate the use of RA in your sysctl by setting in sysctl.conf:

  net.ipv6.conf.all.accept_ra=0
  net.ipv6.conf.default.accept_ra=0

Then, do a "sysctl -p" and manually remove the already assigned route.

The reason why you should not ignore this is:
When you get IPv6 routes via rogue advertisements and your servers is sending 
IPv6 traffic through the attackers server, he will be able to read your traffic.
And - for unencrypted VoIP traffic - he can simply see only all the numbers you 
dialed, seeing what DTMF keys were pressed and finally listen to the voice 
stream.

So - this is definetely worth to investigate and to get your ITSP have a look 
at it. There are many ways to stop other customers from doing this (maybe this 
happens accidently).
If you have further questions you might contact me off-list - since this is 
something that does not really fit in the asterisk list ;-)


Max



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Re: [asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)

[Max Grobecker]

Hi,


Am 28.10.2016 um 17:38 schrieb Markus:
> exten => _-.,1,NoOp(Blocking dash)
> exten => _-.,n,Hangup

> How do I do it right?


why not using FILTER() in your dialplan to eleminate all chars that are not 
numeric?
Like 

   Set(VAR=${FILTER(0-9+),${EXTEN}})

That would eleminate all characters you're not expecting.


Greetings
 Max



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Re: [asterisk-users] Adding a pause when transfering a call

[Max Grobecker]

Hi,

some phones can add a pause when dialing, sometimes by holding the * or # key a 
few seconds after the first digit.
If it works, the phone normally adds a "W" or ";" to the dial string.
So you would program the speed dial key with <*2[hold * or #]101>.

Am 01.10.2016 um 20:22 schrieb Tech Support:
> All;
> 
> When I transfer a call to another extension, I can simply press *2 and 
> then the extension number, say 101. No big deal. The problem I am having is 
> in programming a speed dial key to dial *2101, which is failing. The only 
> thing I can think of is that the speed dial key is dialing the string too 
> fast and Asterisk sees it as <*2101> instead of <*2><101> which fails. How do 
> other people get around this?
> 
> Thanks;
> 
> John  
> 
> 
> 



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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Max Grobecker]

Hi,

OK, then it looks like the client transferred the call anywhere else.
Do you see an entry in your log that refers to the bridge ID 
00bd58c3-3bce-4f1b-9d79-11eb96f37260 ?
If there was a transfer, the call *may* have been bridged with the transfer 
destination. Also, the destination might be external,
so you may see a second call starting at the time where the client left the 
bridge.

Max



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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Max Grobecker]

Maybe the client just put the call on hold.
So the call technically has not ended AND the client does not need to
send or handle any RTP data.
Is there any mention of "music on hold" for this channel?

Greetings
 Max


- Nachricht von Leandro Dardini <ldard...@gmail.com> -
 Datum: Thu, 15 Sep 2016 18:06:14 +0200
   Von: Leandro Dardini <ldard...@gmail.com>
Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
   Betreff: [asterisk-users] Tricking asterisk to think the call has
ended, but it was continuing on the other side
An: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>



I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.

An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.


- Ende der Nachricht von Leandro Dardini <ldard...@gmail.com> -




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[asterisk-users] Get Realtime extension matched entry ID

[Max Grobecker]

Hello,

is there a possibility to get (by dialplan variable?) the entry ID of
the realtime extensions table,
that matched the current call?

For example (simplified):

ID - exten
-
1+49123456
2_+49555.


If I receive a call on +49123456 this surely works with REALTIME_FIELD
and ${EXTEN} as matching field value.
But for ID 2 in ${EXTEN} the full dialed number is stored, so I would
never find a matching field in the database using this way.


Is there any way to get the ID field of the current channel or at
least a variable, where the unexpanded matched "exten" pattern is
stored (i.e. the "_+49555.")?
I just need something unique to find the dialed extension in the table...


Thanks!

Greetings,
 Max


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Re: [asterisk-users] Asterisk 13.11 realtime problem registering phones

[Max Grobecker]

Hi,


Am 02.09.2016 um 22:48 schrieb Carlos Chavez:
> I upgraded my office installation from 13.10 to 13.11 yesterday and now I 
> am having problems registering phones.  Here is what I get on the CLI:
> 
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'qualify_timeout' cannot be type 
> 'int(10)' (need char)
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'expiration_time' cannot be type 
> 'bigint(20)' (need char)
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1246 require_mysql: 
> Possibly unsupported column type 'enum('yes','no')' on column 
> 'authenticate_qualify'
> [Sep  2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: 
> Realtime table general@ps_contacts: column 'via_port' cannot be type 
> 'int(11)' (need char)
> [Sep  2 15:38:46] ERROR[2098]: res_pjsip_registrar.c:411 register_aor_core: 
> Unable to bind contact 
> 'sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525' to 
> AOR '2001'
>   == Contact 
> 2001/sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525 
> has been deleted
> 
> The mysql warnings have always been there since version 13.0 and the 
> "Unable to bind contact..." error has also been present since I started using 
> PJSIP realtime with Asterisk 13 (13.5 at least). 

I hope you find this concerning...

Have you upgraded your MySQL realtime tables to the new schema as introduced 
with Asterisk 13?
-> 
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13#UpgradingtoAsterisk13-RealTime

It's likely a database error (i.e. a required, but missing table field) causes 
this issue.
But even if not, you are getting rid of the warning messages ;-)



Greetings
Max



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Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

[Max Grobecker]

Hi Jonas,


Am 02.09.2016 um 11:26 schrieb Jonas Kellens:
> [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 
> 11.22.33.44:40670
> [Aug 31 14:59:34] -- Now forwarding 
> Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' 
> (thanks to SIP/myaccount184-3729)
> Question : how can I read the variable which contains the value 
> 'myaccount184' in the context from-internal ?

You can get some information out of the REDIRECTING function [1].
For example, your redirecting source (the called device that caused call 
diversion) is normally stored in REDIRECTING(from-num).

[1] https://wiki.asterisk.org/wiki/display/AST/Function_REDIRECTING


Greetings
Max



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Re: [asterisk-users] no silk translation ?

[Max N. Boyarov]

On 11.06.2013, at 0:24, Sean Darcy wrote:

 Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no 
 success:


Silk is enabled only after asterisk restart.

for silk work need codecs.conf with silk configuration
res_format_attr_silk.so - loaded
codec_silk.so - loaded

please see

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats


 
 [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 
 ast_channel_make_compatible_helper: No path to translate from SIP/ng- 
 to Motif/+12025551...@voice.google.com-da3c
 [Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 dial_exec_full: 
 Had to drop call because I couldn't make SIP/ng- compatible with 
 Motif/+12025551...@voice.google.com-da3c
  == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-'
 
 core show translations doesn't include any SILK.
 
 SILK is installed:
 
 core show codec 100018
 100018 SILK Custom Format 8khz
 100018 SILK Custom Format 12khz
 100018 SILK Custom Format 16khz
 100018 SILK Custom Format 24khz
 
 sean
 
 
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[asterisk-users] Help needed for chan_ss7 for Digium device

[Max Alex]

Hi All,
I have installed centos 5.6 32 bit on xeon server and i have also installed
latest version of asterisk 1.6 and dahdi as well.
I want to install chan_ss7 for this server and I want to know about the
following device.
Digium TE420B
I dont know much about the configuration files for Digium TE420B.
Can anybody provide me required ss7.conf file and also provide dahdi
configuration which is needed for this device.
Thanks you so much in advance!!

Thanks,
Max Alex
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[asterisk-users] Look c8m9kl

[Max Alex]

37pgn.
 http://darkskiesblog.com/wp-content/uploads/img/vosc.html
22gv7con3pr fjfvojvm e1cugvfj, tuzj2tcz2 4zssju6k5bfj. jdc52cc
twdoh35sn4s sb2yj.

-- 
Thanks,
Max Alex
Voip Developer

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[asterisk-users] dahdi issue on digium AEX800

[Max Alex]

Hi All,
I have installed asterisk 1.6.2.8
Dahdi: 2.4.0
Digium card:  Digium, Inc. Wildcard AEX800 8-port analog card

I have configured this card properly and it is working for calls too,
But there is issue of one way audio on this outbound routes only,
My voice is going to outside pstn number but i am not able to get their
audio,
I have disabled firewall, selinux is also off.

If I am applying this line to analog phone then also it is working fine,
But when it is added on digium card then this issue happens,
can anybody help me for this issue?

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Dahdi issue on sangoma A200

[Max Alex]

Hi,
Thanks for this information, but it is not working for both the issues,
I have tried with the configuration with cidsignalling, cidstart etc..
Can any one provide more help for this.

Thanks,
Max Alex
Voip Developer



On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru 
beaasteriskg...@gmail.com wrote:

 Hi max,
 Have look on my blog regarding this.


 http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html

 Thanks,
 Ashik

 On Sat, Aug 7, 2010 at 11:15 AM, Max Alex max.aster...@gmail.com wrote:

 Hi All,
 I have Sangoma A200 Card installed on my system,
 I have centos 5.5 with 64 bit,
 Here are the description for asterisk and dahdi.
 Asterisk 1.6..2.9
 Dahdi: 2.3.0.1
 I have two issues with dahdi
 1) I am not getting full callerid on my phones from sangoma card to
 asterisk users. if i am connecting analog phone directly then i am getting
 callerid properly.
 I am in india and using Airtel Connection, I have set variables in
 chan_dahdi.conf as well for callerid but the not getting full digits in
 callerid,
 it is coming with 8 digits only.
 2) Another issue is when I am hanging up the phone from inbound or
 outbound from the dahdi channel, it takes 5-6 seconds to dropping the call.

 Here are the confguration file for chan_dahdi.conf
 -
 ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
 ;autogenrated on 2010-07-30
 ;Dahdi Channels Configurations
 ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

 [trunkgroups]

 [channels]
 context=default
 usecallerid=yes
 callerid=asreceived
 hanguponpolarityswitch=yes
 answeronpolarityswitch=yes
 ;cidstart=ring
 cidstart=polarity_IN
 ;cidsignalling=dtmf
 cidsignalling=dtmf
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 relaxdtmf=yes
 rxgain=0.0
 txgain=0.0
 group=1
 callgroup=1
 pickupgroup=1
 immediate=no
 useincomingcalleridondahditransfer=yes
 ;callerid=asreceived

 ;Sangoma AFT-A200 [slot:4 bus:2 span:1]  wanpipe1
 context=from-internal
 group=1
 echocancel=yes
 callerid=asreceived
 signalling = fxo_ks
 channel = 1

 context=from-internal
 group=1
 echocancel=yes
 callerid=asreceived
 signalling = fxo_ks
 channel = 2

 context=from-zaptel
 group=0
 echocancel=yes
 callerid=asreceived
 signalling = fxs_ks
 channel = 3

 context=from-zaptel
 group=0
 echocancel=yes
 callerid=asreceived
 signalling = fxs_ks
 channel = 4
 ---
 Please hemp me for this issues.

 Thanks,
 Max Alex
 Voip Developer


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[asterisk-users] Dahdi issue on sangoma A200

[Max Alex]

Hi All,
I have Sangoma A200 Card installed on my system,
I have centos 5.5 with 64 bit,
Here are the description for asterisk and dahdi.
Asterisk 1.6..2.9
Dahdi: 2.3.0.1
I have two issues with dahdi
1) I am not getting full callerid on my phones from sangoma card to asterisk
users. if i am connecting analog phone directly then i am getting callerid
properly.
I am in india and using Airtel Connection, I have set variables in
chan_dahdi.conf as well for callerid but the not getting full digits in
callerid,
it is coming with 8 digits only.
2) Another issue is when I am hanging up the phone from inbound or outbound
from the dahdi channel, it takes 5-6 seconds to dropping the call.

Here are the confguration file for chan_dahdi.conf
-
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2010-07-30
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
callerid=asreceived
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
;cidstart=ring
cidstart=polarity_IN
;cidsignalling=dtmf
cidsignalling=dtmf
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
useincomingcalleridondahditransfer=yes
;callerid=asreceived

;Sangoma AFT-A200 [slot:4 bus:2 span:1]  wanpipe1
context=from-internal
group=1
echocancel=yes
callerid=asreceived
signalling = fxo_ks
channel = 1

context=from-internal
group=1
echocancel=yes
callerid=asreceived
signalling = fxo_ks
channel = 2

context=from-zaptel
group=0
echocancel=yes
callerid=asreceived
signalling = fxs_ks
channel = 3

context=from-zaptel
group=0
echocancel=yes
callerid=asreceived
signalling = fxs_ks
channel = 4
---
Please hemp me for this issues.

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Moh help needed

[Max Alex]

Hi All,
I have issue with moh in asterisk 1.4 branch, (1.4.26)
I have created two different moh classes and it is fine,
I have assigned 1st one to sip user 1001, in sip entries like
musiconhold=classname.
done same for 2nd one to sip user 2001,
And now i am making internal calls between then as normal way.
When 1001 calls to 2001 and 2001 answers, After that 2001 will put on hold
to 1001 then it is playing 1001's moh instead of 2001.
I need to configure such a way that whichever user put on hold to another it
can be set and play which is assigned to him.
Like 2nd moh class will be use when 2001 put on hold to 1001.

Please let me suggestions on this.

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Outgoing Calls Only -- Firewall Rules

[Max McGraw]

 Nicholas,

 Sorry I don't know, but are your calls working okay ?

 Depending on the verbosity level being set, I see warning
 msgs all the time, that I ignore.

 Frequently, an upgrade to the next release of the same
 major version also eliminates the warning msgs.

 If you are really concerned, I would find an unused machine,
 install Linux  Asterisk 1.6.x on it, try out your calls and
 see if the warnings still appear.

 If there are no warnings of this kind, it is an issue specific
 to a module in that 1.4.x release and likely to go away.

 Good luck !

--

  On Tue, Jan 5, 2010,   Nicholas Blasgenwrote:

 Asterisk 1.4.29 or so.

 access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range
 1 2
 access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq
 5060

 But yes, all your feedback worked.  I didn't need to port-forward any
 incoming ports, only 5060/1-2 for outgoing UDP.  The only issue I'm
 now having is:

 --- SIP read from 66.227.100.20:5060 ---
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 209.34.93.68:5060;branch=z9hG4bK3eb38bde;rport=51566
 
 Warning: 392 66.227.100.20:5060 Noisy feedback tells:  pid=9611
 req_src_ip=209.34.93.68 req_src_port=51566 in_uri=sip:sip.jnctn.net
 out_uri=sip:sip.jnctn.net via_cnt==1

 209.34.93.68 is my IP, 209.34.93.68 is Junction Networks (for this
 example).  I also get it from my backbone providers as well so it's likely
 something to do with that 51566 req_src_port thing.  Any idea what this is
 an how to configure it to a restricted range of IP addresses?

 Nicholas Blasgen
 Partner / Network Operations
 Refractive Dialer LLC
 (724) 252-7436


 On Sun, Jan 3, 2010 at 8:29 PM,  Max McGraw   wrote:

  Nicholas,

  you haven't specified which version, which does make
  a lot of difference.

  1.6.x  can easily traverse NAT. If you are only making
  outbound calls, you shouldn't need to forward 5060.

  Unless you have a special NAT that is blocking
  outbound connections, the  SIP.conf  settings below
  should work whether your provider uses SIP
  registrations or not. My codec related settings may
  not be applicable to your installation :

  ; -
  [general]
  dtmfmode=rfc2833
  relaxdtmf=yess
  bandwidth=high
  disallow=all
  allow=ulaw
  ;
  ;   NAT stuff
  ;
  localnet=192.168.x.0/255.255.255.0
  externip=a.b.c.d:5060
  nat=yes
  ;
  ;   Media stuff
  ;
  canreinvite=no
  ;
  ;
  [your-voip-provider-para]
  ;
  context=default
  type=friend
  ;
  ;  your provider's outbound gateway
  ;
  host=w.x.y.z
  ;
  dtmfmode=rfc2833
  relaxdtmf=yess
  disallow=all
  allow=ulaw
  ;
  ; -


  On Sun, Jan 3, 2010,   Nicholas Blasgen    wrote:

  I'm trying to move my Asterisk deployments under a Virtual IP address
  and
  now remember why I dislike this.  My primary Asterisk system is now
  behind a
  firewall in private address space.  My question is what ports are needed
  to
  be opened just for the purpose of placing outgoing calls.  I would have
  assumed none, but I can't even get replies on registration from any of
  my 3
  VoIP providers.  I tried defining the External IP and some other stuff,
  but
  I assume it's fully an issue with the firewall.  Do I really need 5060
  port
  forwarded just to register with remote hosts?
 
  Nicholas Blasgen
  Partner / Network Operations
  Refractive Dialer LLC
  (724) 252-7436
 
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[asterisk-users] Canadian call quality issue

[Max McGraw]

 hello,

 we have been using a couple of US based
 VoIP providers for outbound calls completed
 within the US, without any issues.

 We recently started making calls to Canada
 and have received a few complaints about
 the call quality.

 Questions :

  - Could this be because of the number of
intermediate IP hops between us / our
VoIP provider and the Canadian phone
companies ?

  - Would choosing a Canadian VoIP provider
address / resolve this issue ?

 Thank you in advance.

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Re: [asterisk-users] Really Silly Question From Total Newbie

[Max McGraw]

 On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

 wow !  are you serious ?

 you can afford an iPhone but your entire project is
 dead in the water over a couple of bucks.

 I know there is a contradiction somewhere in there.

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Re: [asterisk-users] Canadian call quality issue

[Max McGraw]

 Jon/Kyle,

 thank you for the feedback.

 I checked with someone who manages a much
 higher volume of calls to Canada and he said
 there are some pockets  some providers that
 report issues with call quality. Overall the calls
 sound the same as they do in the US.

--

  On Tue, Jan 5, 2010,  jon pounder   wrote:

 Kyle Kienapfel wrote:
 Going along the internet between us and canada doesn't add much
 distance, but bouncing back and forth between east and west coast
 does.


 I'm not so sure that is the case, what I do know is both Rogers and Shaw
 can never seem to fix complaint issues with voip unless you are using
 their phone service. We just gave up on it and I will not ever spend a
 penny with Rogers as a result since I am convinced they are deliberately
 filtering things so you are locked into their voice services.

 Other than that, voip works just fine.

 On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:

  hello,

  we have been using a couple of US based
  VoIP providers for outbound calls completed
  within the US, without any issues.

  We recently started making calls to Canada
  and have received a few complaints about
  the call quality.

  Questions :

  - Could this be because of the number of
    intermediate IP hops between us / our
    VoIP provider and the Canadian phone
    companies ?

  - Would choosing a Canadian VoIP provider
    address / resolve this issue ?

  Thank you in advance.

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Re: [asterisk-users] Really Silly Question From Total Newbie

[Max McGraw]

 my apologies, I do understand.

 sorry.

--

  On Tue, Jan 5, 2010,  UIT DEV  wrote:

 Yep.  Its called unemployment.   Got the iPhone a little less than a
 year ago.   Someone in India got my job in mid-November.   I got stuck
 holding the 2-year contract.    Oh well.   Such is life.

 Look - I am going to retire from this thread.   Everyone's been a
 great help and I know you and others dont know my situation and I am
 not one to broadcast it - but when prodded.  there it is.
 Yes, several bucks leads to more than several bucks and being
 unemployed and living off the wife's income - its not an option.
 Hopefully you'll not encounter sucky times, else you'd know..    That
 couple of bucks a month will never just be a couple of bucks..  :-)


 On Tue, Jan 5, 2010 at 4:47 PM, Max McGraw max.mcg...@gmail.com wrote:
  On Tue, Jan 5, 2010,   UIT DEV   wrote:

 Steve-

 Got an iPhone  [...]

 As I got to reading I began to see things like provider, as you've
 said here, and unfortunately if that is the only way then I will have
 to stop here as I do not have funds to further this little experiment.
  I guess I was under the impression that with a lot of configuring and
 reading and technical assistance, etc - I could create what I suppose
 the VoIP provider is basically doing, trying to avoid yet another
 monthly charge.  [...]

  wow !  are you serious ?

  you can afford an iPhone but your entire project is
  dead in the water over a couple of bucks.

  I know there is a contradiction somewhere in there.

 --

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Re: [asterisk-users] Outgoing Calls Only -- Firewall Rules

[Max McGraw]

 Nicholas,

 you haven't specified which version, which does make
 a lot of difference.

 1.6.x  can easily traverse NAT. If you are only making
 outbound calls, you shouldn't need to forward 5060.

 Unless you have a special NAT that is blocking
 outbound connections, the  SIP.conf  settings below
 should work whether your provider uses SIP
 registrations or not. My codec related settings may
 not be applicable to your installation :

 ; -
 [general]
 dtmfmode=rfc2833
 relaxdtmf=yess
 bandwidth=high
 disallow=all
 allow=ulaw
 ;
 ;   NAT stuff
 ;
 localnet=192.168.x.0/255.255.255.0
 externip=a.b.c.d:5060
 nat=yes
 ;
 ;   Media stuff
 ;
 canreinvite=no
 ;
 ;
 [your-voip-provider-para]
 ;
 context=default
 type=friend
 ;
 ;  your provider's outbound gateway
 ;
 host=w.x.y.z
 ;
 dtmfmode=rfc2833
 relaxdtmf=yess
 disallow=all
 allow=ulaw
 ;
 ; -


  On Sun, Jan 3, 2010,   Nicholas Blasgenwrote:

 I'm trying to move my Asterisk deployments under a Virtual IP address and
 now remember why I dislike this.  My primary Asterisk system is now behind a
 firewall in private address space.  My question is what ports are needed to
 be opened just for the purpose of placing outgoing calls.  I would have
 assumed none, but I can't even get replies on registration from any of my 3
 VoIP providers.  I tried defining the External IP and some other stuff, but
 I assume it's fully an issue with the firewall.  Do I really need 5060 port
 forwarded just to register with remote hosts?

 Nicholas Blasgen
 Partner / Network Operations
 Refractive Dialer LLC
 (724) 252-7436

 __

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[asterisk-users] DeadAgi application issue

[Max Alex]

Hi All,
I have working asterisk 1.4.24.1, but I have issues with DeadAgi
application.
I am using hylafax and iaxmodem with asterisk, mail 2 fax and fax 2 mail
feature.

My system details are below:
OS: Centos 5.3
Asterisk Version: 1.4.24.1
Dahdi version:  dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2
Zap device: Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1
AFT card
Kernel: 2.6.18-128.1.10

We have used HylaFAX+ for the mail 2 fax and fax2mail feature.
We have setup incoming and outgoing phpagi scripts for the calculation for
fax and billing too.
But when we are sending and receiving the faxes it is working fine,
But some times the deadagi application got stuck the channels.
And becuase of that the phpagi scripts are not completed.
In that case we need to restart asterisk complusary.
We are using zap lines for the outbound and inbound faxes.

Can any one suggest solution for this?

Thanks in advance!!!

Thanks,
Max Alex
Voip Developer
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[asterisk-users] 2BCT last mile... Hopefully

[Max Metral]

Ok, so I've made progress on 2BCT (2 B-Channel Transfer).  I'm assuming that
the debug info below shows that XO doesn't have 2BCT enabled on my line, but
if anybody can confirm that'll let me be way more indignant. J

 

-- Native bridging DAHDI/1-1 and DAHDI/3-1

 Protocol Discriminator: Q.931 (8)  len=28

 Call Ref: len= 2 (reference 801/0x321) (Terminator)

 Message type: FACILITY (98)

 [1c 15 91 a1 12 02 01 23 06 07 2a 86 48 ce 15 00 08 30 04 02 02 01 93]

 Facility (len=23, codeset=0) [ 0x91, 0xA1, 0x12, 0x02, 0x01, '#', 0x06,
0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x04, 0x02, 0x02, 0x01,
0x93 ]

PROTOCOL 11I

A1 0012 (CONTEXT SPECIFIC [1])

  02 0001 23 (INTEGER: 35)

  06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08)

  30 0004 (SEQUENCE)

02 0002 01 93 (INTEGER: 403)

 Protocol Discriminator: Q.931 (8)  len=5

 Call Ref: len= 2 (reference 801/0x321) (Originator)

 Message type: CONNECT ACKNOWLEDGE (15)

q931.c:3705 q931_receive: call 801 on channel 1 enters state 10 (Active)

 Protocol Discriminator: Q.931 (8)  len=16

 Call Ref: len= 2 (reference 801/0x321) (Originator)

 Message type: FACILITY (98)

 [1c 09 91 a3 06 02 01 23 02 01 00]

 Facility (len=11, codeset=0) [ 0x91, 0xA3, 0x06, 0x02, 0x01, '#', 0x02,
0x01, 0x00 ]

PROTOCOL 11I

A3 0006 (CONTEXT SPECIFIC [3])

  02 0001 23 (INTEGER: 35)

  02 0001 00 (INTEGER: 0)

-- Processing IE 28 (cs0, Facility)

Handle Q.932 ROSE return error component

Unable to handle return result on switchtype 1!

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[asterisk-users] 2B Channel Transfer on XO-based T1

[Max Metral]

I'm trying to get blind transfer from an incoming DAHDI line to an
external number to work on an * 1.6 install using a T1 from XO.  The
documentation is very distributed and incomplete, so while it's not
working, it's definitely more likely my error somehow.  Couple questions if
anybody is out there who even knows what TBCT is.

 

1)  Is this even supported?

2)  Does it require some settings in dahdi_channels, or features, or
whatever?

3)  Would I trigger it via a Dial command or commands, or via
Transfer?

4)  Do either or both of the legs need to be answered?

 

Thanks very much.

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Re: [asterisk-users] [asterisk-dev] Grandstream blind transfer issue

[Max Alex]

Hi All,
Thanks for your suggestions.
I am using DeadAgi application for origination of calls, i have set same
context to Transfer context.
I have also added Tt options in dial options.
When I am receiving calls to grandstream phone,
I am using transfer button to transfer the call, but it is not transfering
with AGI application,
Can anyone provides me suggestions for blind transfer with AGI application.

My Dialplan is given Below. I have used PHPAGI for the origination of calls.
[bt200]
exten = _X.,1,Set(__TRANSFER_CONTEXT=bt200)
exten = _X.,n,DeadAGI(testing_agi/testing.php)
exten= h,1,NoOp(${DIALSTATUS})


Thanks,
Max Alex
Voip Developer



On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion klaus.mailingli...@pernau.at
 wrote:

 Haven't you read my email?

 1. Wrong list
 2. Missing log entries (set debug 4, set verbose 4)

 klaus

 Max Alex schrieb:
  Hi All,
  Thanks for your reply.
  I got this refer message in asterisk.
  but there is not any active channel of blind transfer.
  --
  REFER sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 mailto:
 sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25 SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
  From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
  To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25
  mailto:sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
 ;tag=as32ed6c48
  Contact: sip:7...@192.168.1.30:5060;transport=udp
  Supported: replaces, path
  Refer-To: sip:1631...@192.168.1.25 sip%3a1631...@192.168.1.25
  mailto:sip%3a1631...@192.168.1.25sip%253a1631...@192.168.1.25
 
  Referred-By: sip:7...@192.168.1.25 sip%3a7...@192.168.1.25 mailto:
 sip%3a7...@192.168.1.25 sip%253a7...@192.168.1.25
  Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
  mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
  CSeq: 34526 REFER
  User-Agent: Grandstream BT200 1.1.6.46
  Max-Forwards: 70
  Allow:
  INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
  Content-Length: 0
 
  -
  --- (14 headers 0 lines) ---
  Call 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
  mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 got a SIP call
  transfer from caller: (REFER)!
  SIP transfer to extension 1631...@outgoing by 7...@192.168.1.25
  mailto:7...@192.168.1.25
  localhost*CLI
  --- Transmitting (NAT) to 192.168.1.30:5060 http://192.168.1.30:5060
 ---
  SIP/2.0 202 Accepted
  Via: SIP/2.0/UDP
  192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30
  From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
  To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25
  mailto:sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
 ;tag=as32ed6c48
  Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
  mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
  CSeq: 34526 REFER
  User-Agent: Asterisk PBX
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  Supported: replaces
  Contact: sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 mailto:
 sip%3a1...@192.168.1.25 sip%253a1...@192.168.1.25
  Content-Length: 0
 
 
  
  
  Is there any options we need to enable in asterisk or grandstream phone?
  I have already user transfer option 'Tt' in dialplan of this.
  Please provide me some help.
  Thanks in advance!!
 
  Thanks,
  Max Alex
  Voip Developer
 
 
 
  On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion
  klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at
 wrote:
 
  Max Alex wrote:
Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
 
  Does it work with other phones? That means is it a Grandstream isue
 or a
  general issue?
 
I have updated the latest firmware to the phone.
The phone is sending the *refer* to asterisk but asterisk is not
  able to
connect with the another call
 
  Why? some log messages would help us helping you.
 
that i have checked in sip debug.
I am using transfer button of the grandstream phone.
Can anybody provide help for this issue?
 
  Please ask again on the user mailing lists and provide some log
 messages
 
Thanks in advance!!
   
Thanks,
Max Alex
Voip Developer
   
   
   
 
 
   
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Re: [asterisk-users] [asterisk-dev] Grandstream blind transfer issue

[Max Alex]

Hi All,
Thanks for your reply.
I got this refer message in asterisk.
but there is not any active channel of blind transfer.
--
REFER sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25;tag=as32ed6c48
Contact: sip:7...@192.168.1.30:5060;transport=udp
Supported: replaces, path
Refer-To: sip:1631...@192.168.1.25 sip%3a1631...@192.168.1.25
Referred-By: sip:7...@192.168.1.25 sip%3a7...@192.168.1.25
Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
CSeq: 34526 REFER
User-Agent: Grandstream BT200 1.1.6.46
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

-
--- (14 headers 0 lines) ---
Call 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 got a SIP call transfer
from caller: (REFER)!
SIP transfer to extension 1631...@outgoing by 7...@192.168.1.25
localhost*CLI
--- Transmitting (NAT) to 192.168.1.30:5060 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.30:5060
;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30
From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687
To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25;tag=as32ed6c48
Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25
CSeq: 34526 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:1...@192.168.1.25 sip%3a1...@192.168.1.25
Content-Length: 0




Is there any options we need to enable in asterisk or grandstream phone?
I have already user transfer option 'Tt' in dialplan of this.
Please provide me some help.
Thanks in advance!!

Thanks,
Max Alex
Voip Developer



On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion klaus.mailingli...@pernau.at
 wrote:

 Max Alex wrote:
  Hi All,
  I have working asterisk version 1.4.24.
  I have a blind transfer issue with grandstream bt200.

 Does it work with other phones? That means is it a Grandstream isue or a
 general issue?

  I have updated the latest firmware to the phone.
  The phone is sending the *refer* to asterisk but asterisk is not able to
  connect with the another call

 Why? some log messages would help us helping you.

  that i have checked in sip debug.
  I am using transfer button of the grandstream phone.
  Can anybody provide help for this issue?

 Please ask again on the user mailing lists and provide some log messages

  Thanks in advance!!
 
  Thanks,
  Max Alex
  Voip Developer
 
 
  
 
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Re: [asterisk-users] Grandstream blind transfer issue

[Max Alex]

Hi
I have used the transfer operation this way.
When i got a call on grandstream phone, i will receive it
and press transfer button and enter transfer number and press send button.
My call is disconnected but no call transfer from asterisk.
Please advice me!!
Thanks,
Max Alex
Voip Developer



On Tue, Apr 7, 2009 at 11:12 PM, Gordon Henderson 
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:

 On Tue, 7 Apr 2009, Max Alex wrote:

  Hi All,
  I have working asterisk version 1.4.24.
  I have a blind transfer issue with grandstream bt200.
  I have updated the latest firmware to the phone.
  The phone is sending the *refer* to asterisk but asterisk is not able to
  connect with the another call
  that i have checked in sip debug.
  I am using transfer button of the grandstream phone.
  Can anybody provide help for this issue?
  Thanks in advance!!

 How are you doing the entire transfer operation?

 For blind transfers, I do:

  Push Transfer
 (caller is now on hold, you get a new dial-tone)
   dial extension and push SEND

 At this point, called phone rings and caller is immediately taken off hold
 and transfered to the new ringing phone... you can hang up at that point.

 Don't use the 'flash' key.

 I have many BT200's and GXP280's out there - this seems to work for them
 without any issues. Asterisk 1.2 though.


 Gordon



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[asterisk-users] Grandstream blind transfer issue

[Max Alex]

Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
I have updated the latest firmware to the phone.
The phone is sending the *refer* to asterisk but asterisk is not able to
connect with the another call
that i have checked in sip debug.
I am using transfer button of the grandstream phone.
Can anybody provide help for this issue?
Thanks in advance!!

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Asterisk crashed!!!

[Max Alex]

Hi All,
I have a working asterisk 1.4.23.1 on server.
OS: Centos 5.2
Suddenly asterisk has stopped to process calls  crashed.
I found that asterisk has generated coredumps.
I have restarted asterisk  it started to work as expected without any
issue.
Would you please help me out to troubleshoot the cause of crash?
Please checkout following link, I have uploaded coredump backtraces there.
http://pastebin.com/m5480bcb8

Please provide me help regarding this.
Thanks in advance.

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Hold/Resume issue with polycom

[Max Alex]

Hi All,
We have working pbx with asterisk 1.4.23.1
System: Centos 5.2
We are using polycom phones for pbx.
We are using sip channels for calls and all the users has set canreinvite=no
and nat=yes.
We have a issue with resuming the hold call by the polycom phone when the
call traffic is high.
We have put the incoming call on hold and when we try to resume it back,
the call is hangup, and not able to connect the hold channel.
Can anyone provide help!!

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Need help on Forwarding

[Max Alex]

Hi All,
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have set one of the extension in call forwarding
in phone (linksys)
then it says to me 302 Moved Temporarily and call is forwarded to that
number.
In this i need to disable the forwarding from dialplan or any configuration
method, so when the ring group is in process then no call will be forwarded.
Please provide help regarding this!!
Thanks in Advance!!
Thanks,
Max Alex
Voip Developer
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[asterisk-users] DeadAgi Application in asterisk 1.6

[Max Alex]

Hi All,
I have configured the phpagi application for counting the duration of call,
The call is originated from the script and after hangup the call the
duration and status will be stored.
This functionality and php script is working fine with deadagi application
with asterisk 1.4.
I have a problem with asterisk 1.6 deadagi application, when the call is
hangup at that time the script is exited and no duration and status will be
counted, So please provide help regarding this deadagi application in
asterisk 1.6 branch,
Please help me regarding this!!
Thanks in Advance!!!

Thanks,
Max Alex
Voip Developer
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[asterisk-users] EVRC support

[Max Alex]

Hi All,
I am working with asterisk 1.4 branch
I need to know whether EVRC codec works with asterisk version or not?
If caller and callee both has EVRC support then how the asterisk will
transmit the audio with this codecs.
I need to know the working role of asterisk with EVRC while it is running.

Please provide information!!!
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] I need help

[Max Brooks]

Bayardo Sanchez wrote:
 i have a problem need help

 == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 
 'SIP/8022-b7225740'
 -- Got SIP response 503 Service Unavailable back from 74.63.41.218
 -- SIP/voipms4-09ab0c38 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)
   == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION'
   == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero 
 on 'SIP/8010-b72241b0'

if you have made no changes to your asterisk configuration since it last 
worked Contact your service provider as they have an issue more likely 
though is you have a bad dial command.

/telepathy

-- 
Kind Regards
Max Brooks - Developer

Legatio Technologies Limited

Phone: 01793 520 506

www.legatio.com, www.ftax.co.uk

Legatio is part of the Callcredit Information Group: www.skipton.com, 
www.callcredit.co.uk, www.eurodirect.co.uk

Legatio Technologies Limited,
One Park Lane,
Leeds,
West Yorkshire,
LS3 1EP
Registered in England and Wales No. 4519902

-

Powered by Legatio.com

This message is confidential. It may not be disclosed to, or used by, anyone 
other than the addressee(s). If you receive this message in error, please 
advise us immediately using the email address i...@legatio.com. Internet e-mail 
is not necessarily secure. Legatio will not accept responsibility for 
alterations or additions to any e-mail message or attached documents that occur 
after transmission.


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Re: [asterisk-users] Root Password not taking

[Max Brooks]

Jim Dickenson wrote:
 What I have done in the past to set the password for root is to boot 
 in rescue mode and edit /etc/shadow setting the password to some know 
 value from another system.
 -- 
 Jim Dickenson
 mailto:dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/

I personally prefer to chroot into the / partition and run passwd.

-- 
Kind Regards
Max Brooks - Developer

Legatio Technologies Limited

Phone: 01793 520 506

www.legatio.com, www.ftax.co.uk

Legatio is part of the Callcredit Information Group: www.skipton.com, 
www.callcredit.co.uk, www.eurodirect.co.uk

Legatio Technologies Limited,
One Park Lane,
Leeds,
West Yorkshire,
LS3 1EP
Registered in England and Wales No. 4519902

-

Powered by Legatio.com

This message is confidential. It may not be disclosed to, or used by, anyone 
other than the addressee(s). If you receive this message in error, please 
advise us immediately using the email address i...@legatio.com. Internet e-mail 
is not necessarily secure. Legatio will not accept responsibility for 
alterations or additions to any e-mail message or attached documents that occur 
after transmission.


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[asterisk-users] Realtime MOH

[Max Alex]

Hi All,
I have set up realtime configuration of asterisk with mysql,
and it is working fine.
Asterisk version is :1.4.21
I have a issue regarding MOH, i have created musiconhold.conf in database as
per custom configuration.
When we reload moh then it is working fine, but some times the moh get
disappered and we must have to reload to load moh again.

Can any body please help me regarding MOH configuration!!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Local channel Help required

[Max Alex]

Hi,
Thanks for your reply.
I have already used this
exten= 1002,1,Dial(SIP/1002|30|rg)
exten= 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)

but my incoming call is getting hangup, it is not going to second priority.
So is there any configuration we have to do in local channel.

Thanks,
Max Alex
Voip Developer



On Mon, Jan 12, 2009 at 6:45 PM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:

 Philipp Kempgen schrieb:
  Max Alex schrieb:
 
  If i got the NOANSWER then the channel is not passing to next priority.
  I need to pass that channel to the next priority of the context
  [macro-mypbx] so i can set voicemail there.
 
  I want to know how can we set the local channel to go in next priority
 in
  case of NO ANSWER.
 
  core show application Dial
  ---cut---
  g- Proceed with dialplan execution at the current extension if
 the
 destination channel hangs up.
  ---cut---

 No, wait, you don't need the g option here. Sorry.

 Dial() continues after ${DIALSTATUS} = NOANSWER anyway.

 Dial(SIP/${EXTEN});
 if (${DIALSTATUS} = NOANSWER) {
// go to voicemail
 }


   Philipp Kempgen

 --
 AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
 AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 --

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Re: [asterisk-users] Local channel Help required

[Max Alex]

Hi All,
We have already use 'g' option in that, but it is not working in my case.
Thanks,
Max Alex
Voip Developer



On Sun, Jan 11, 2009 at 1:51 AM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:

 Max Alex schrieb:

  If i got the NOANSWER then the channel is not passing to next priority.
  I need to pass that channel to the next priority of the context
  [macro-mypbx] so i can set voicemail there.
 
  I want to know how can we set the local channel to go in next priority in
  case of NO ANSWER.

 core show application Dial
 ---cut---
g- Proceed with dialplan execution at the current extension if the
   destination channel hangs up.
 ---cut---

   Philipp Kempgen

 --
 AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
 AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 --

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[asterisk-users] Local channel Help required

[Max Alex]

Hi All,
I am using asterisk 1.4 branch on server.
Here is a my dialplan.
i have set the incoming route to incoming context, and then i have set dial
with local channel,
The call comes to my server and the call is routed to matched case, so my
phone 1001 ring for 30 seconds.
If i got the NOANSWER then the channel is not passing to next priority.
I need to pass that channel to the next priority of the context
[macro-mypbx] so i can set voicemail there.

I want to know how can we set the local channel to go in next priority in
case of NO ANSWER.

[incoming]
exten= _X.,1,Dial(Local/${ext...@pbx_tech/n)
exten= _X.,2,NoOp(Test)

[pbx_tech]
exten=_X.,1,Macro(mypbx)

[macro-mypbx]
exten= 1001,1,Dial(SIP/1001|30|rg)
exten= 1001,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1001)
exten= 1002,1,Dial(SIP/1002|30|rg)
exten= 1002,2,ExecIf($['${DIALSTATUS}'!='ANSWER']|Macro|voicedid|1002)
exten=s,1,Goto(${MACRO_EXTEN}|1)

[macro-voicedid]
exten=s,1,NoOp(${ARG1})

Please provide me help regarding this!!!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Asterisk CLI got freezed!!

[Max Alex]

Hello,
Thanks for your reply!
I want to confirm that any other things can cause this freeze issue or not,
and how can we prevent this such case.
If asterisk got freeze regarding the down connection time with dns server,
but when it is able to access then asterisk will resolve this freeze issue
itself or we have to setup some preventions for that.

Can anybody suggest me regarding this freeze cli issue!
Thanks in advance!!
Thanks,
Max Alex
Voip Developer



On Wed, Jan 7, 2009 at 7:07 PM, Grygoriy Dobrovolskyy
megaho...@gmail.comwrote:



 2009/1/7 Max Alex max.aster...@gmail.com

 Hi,
 Thanks for your reply
 Can you suggest me how can we avoid it by doing any configuration changes
 in asterisk.
 So the freeze issue may not be occurred again!
 Please provide me some help!!!
 Thanks in advance!

 Thanks,
 Max Alex
 Voip Developer



 On Wed, Jan 7, 2009 at 12:58 PM, Grey Man greymanv...@gmail.com wrote:

 Doesn't matter if you have set it up or not Asterisk needs DNS. I
 haven't checked the code but I think it even does reverse lookups on
 IP addresses. If you haven't got a reliable DNS server available for
 Asterisk I suspect you're always going to get issues.

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 You can setup a local dns server.

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Re: [asterisk-users] Asterisk CLI got freezed!!

[Max Alex]

Hi,
Thanks for your reply
Can you suggest me how can we avoid it by doing any configuration changes in
asterisk.
So the freeze issue may not be occurred again!
Please provide me some help!!!
Thanks in advance!

Thanks,
Max Alex
Voip Developer



On Wed, Jan 7, 2009 at 12:58 PM, Grey Man greymanv...@gmail.com wrote:

 Doesn't matter if you have set it up or not Asterisk needs DNS. I
 haven't checked the code but I think it even does reverse lookups on
 IP addresses. If you haven't got a reliable DNS server available for
 Asterisk I suspect you're always going to get issues.

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[asterisk-users] Asterisk CLI got freezed!!

[Max Alex]

Hi All,
I am using asterisk 1.4.21  with iaxmodem and  hylafax which is sending fax
from my system with zap device.
I am facing a problem that some times my asterisk CLI got freeze and i am
not able to get any information from asterisk.
I need to restart the asterisk compulsory to work it again.
And because of this my iaxmodems are also getting time out from asterisk.

Please provide some help regarding this freeze issue.

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Asterisk CLI got freezed!!

[Max Alex]

HI,
Thanks for your reply,
But we have not setup DNS servers in asterisk. Asterisk is not getting any
DNS requests.
Please provide help regarding this.
Thanks,
Max Alex
Voip Developer



On Tue, Jan 6, 2009 at 4:10 PM, Grey Man greymanv...@gmail.com wrote:

 Make sure the DNS servers Asterisk is using are not becoming
 unresponsive or unreachable. Asterisk blocks on DNS requests so if it
 doesn't get a response it will appear frozen.

 Regards,

 Greyman.

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[asterisk-users] Need help for transfer

[Max Alex]

Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with forwarding enable to 2103.
But is there any procedure in asterisk that we can not allow 2102 not to
forward on 2103.
and also i want to prevent the SIP/2.0 302 Moved Temporarily.
please advice me that how can we set the user for not to forward or transfer
on 2103.
i have tested with allowtransfer=no in sip.

Thanks in advance!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Anonymous callerid

[Max Alex]

Hi,
Thanks for your reply.
Actually we are setting up the callerid in case of emergency calls when we
got the anonymous callerid from the caller.
But the calls are going with callerid anonymous and not set the callerid, i
want to know how can we sent some meaning ful information to the emergency
services so our calls will not be disconnected and recieved by them.
Please provide some help for this.


Thanks,
Max Alex
Voip Developer



On Sun, Nov 30, 2008 at 1:07 AM, Philipp Kempgen
[EMAIL PROTECTED]wrote:

 Max Alex schrieb:

  Actully we are getting the anonymous callerid from the originated phone
  (blocked from phone) so we need to override the callerid and then pass to
  network.
  we need to send out caller id. That is why we tried to override it.
 
  But we are not able to override it.

 I don't quite understand the problem.
 The phone sends anonymous and you want to send something meaningful
 to emergeny services?
 Set(CALLERID(num)=yournumber)
 Set(CALLERID(name)=yournumber)

  On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen
  [EMAIL PROTECTED]wrote:
 
  Max Alex schrieb:
 
   I have one issue regarding override callerid when i have anonymous
 call.
   I have added PAI in sip header and also set sendrpid = yes in sip.conf
   but the callerid is not overriding while i am sending call to three
 digit
   calling like 911.
 
  The caller ID sent to emergency or law enforcement numbers is
  network-provided not user-provided so you can't override it.

   Philipp Kempgen

 --
 http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
 Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Anonymous callerid

[Max Alex]

Hi
Thanks for your reply.
Actully we are getting the anonymous callerid from the originated phone
(blocked from phone) so we need to override the callerid and then pass to
network.
we need to send out caller id. That is why we tried to override it.

But we are not able to override it.
Please help for this!


Thanks,
Max Alex
Voip Developer



On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen
[EMAIL PROTECTED]wrote:

 Max Alex schrieb:

  I have one issue regarding override callerid when i have anonymous call.
  I have added PAI in sip header and also set sendrpid = yes in sip.conf
  but the callerid is not overriding while i am sending call to three digit
  calling like 911.

 The caller ID sent to emergency or law enforcement numbers is
 network-provided not user-provided so you can't override it.

   Philipp Kempgen

 --
 http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
 Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 --

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[asterisk-users] Disable Transfer

[Max Alex]

Hi All,
I want to prevent transfer on based of user,
means we can disable any user or peer to transfer calls in asterisk.
Can any one helps how can we prevent transfer feature.
I am using asterisk 1.4 branch.

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Anonymous callerid

[Max Alex]

Hi All
I have one issue regarding override callerid when i have anonymous call.
I have added PAI in sip header and also set sendrpid = yes in sip.conf
but the callerid is not overriding while i am sending call to three digit
calling like 911.
please give some idea and help for this issue!
I am using asterisk 1.4 branch.

thanks in advance!!
Thanks,
Max Alex
Voip Developer
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[asterisk-users] Asterisk Instant message passing with eyebeam

[Max Alex]

Hi All,
I am searching about asterisk IM message passing with eyebeam.
but i am not able to send instant message to another registered users.
i am working in asterisk 1.4 branch.
i have tested within call and without call but there is no message recieved.
and every time i got error user not found in eye beam.
and in asterisk i got Method is not implemented.

Can anybody helps me in this?
If any patches are there then please let me know.
Thanks in advance!!
Thanks,
Max Alex
Voip Developer
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[asterisk-users] RTP LOG

[Max Alex]

Hi All,
I am using asterisk 1.4.22 in my local system
I want to know how can we set ability to log and report RTP and jitter
statistics per call.
Is there any configuration in logger or configuration in rtp?
Please provide some guide lines for this.
Thanks in advance!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] RTP LOG

[Max Alex]

Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in advance.

Thanks,
Max Alex
Voip Developer



On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen
[EMAIL PROTECTED][EMAIL PROTECTED]
 wrote:

 Positively Optimistic [EMAIL PROTECTED] writes:

  exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
  exten = h,2,Hangup()
  results in
  Set(SIP/rpx2399a-b61fc5e0,
 
 CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00)

 Does it still only report what was in the last incoming RTCP packet?


 /Benny


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[asterisk-users] SRTP support in asterisk 1.6

[Max Alex]

Hi All,
I am checking srtp support in asterisk 1.6,
Let me know any patches available or changes needed for srtp support in
asterisk 1.6.
Thanks in advance!

Thanks,
Max Alex
Voip Developer
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[asterisk-users] changing from default codec

[Max McGraw]

hello,

 I am using sip, my default codec is set to gsm in sip.conf

 Using call files, is there a way to send out a call using
 ulaw while other channels are using gsm ?

 tia.

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[asterisk-users] changing from default codec

[Max McGraw]

hi, using sip, my default codec is set to gsm in sip.conf

I occasionally want to send out a call using ulaw while other channels
are using gsm, how can I do this using call files ?

I couldn't find any codec parameter in the call file definition.

tia.

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[asterisk-users] catch the use of h option in dial

[Max Börebäck]

I like to  determine that the called user pressed ** to disconnect the call (h 
option for Dial CMD ), and not just hang up the phone. 

Is there a way to get that information? 

The context file where the call file connects the call person is included (it 
is simplified ). 
First thing in the context is calling up the destination, the destination ends 
up in a IVR to accept or decline the call. (The same can be done with privacy 
option as I under stand.) 

When the call is connected I like the destination via h option to be able to 
press ** to disconnect and I like to catch this to log that the destination did 
this and not just hang up the phone when the conversation finished. 

Is there a way to do that? 

[incoming] 
exten = s,1,Wait(1) 
exten = s,n,Answer() 
exten = s,n,Dial(Local/${ DSTPHONE [EMAIL PROTECTED]|60| trgM ( dst -ivr^${ 
CALLER_NAME })) 
exten = s,n, Noop (After Dial) 
; Destination hangup 
exten = s,n, Goto (result-${ PC_STATUS }|10) 
;No we get here when destination hangup, both normal hangup and by pressing ** 
; We get here if the dst accepted the call and then pressed ** to abuse 
exten = result-1,10,Playback(custom/ callrejected ) 
exten = result-1,n,Hangup() 
exten = h,1, Noop (PC - Hangup) 
; HERE I LIKE TO LOG if destination pressed ** or just hangup 
exten = h,n,system(/ pc /bin/ log_call ${ DIALEDTIME }:::${ ANSWEREDTIME 
}:::${ PC_STATUS }:::${ HANGUPCAUSE }:::${ DIALSTATUS } ) 

Regards Max 
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[asterisk-users] Asterisk Callerid Help Needed

[Max Alex]

Hi All,
I need some help to about override callerid,
if i get blocked callerid and also having privacy=full.
i am trying to override callerid on that call, but the callerid is not
changed

The sip trace is given below
 INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-23a4ba1;rport
From: Anonymous sip:[EMAIL PROTECTED];tag=89cc6491fcf8ae21o1
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous sip:[EMAIL PROTECTED]:5061
Expires: 240
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 308
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, x-sipura
Content-Type: application/sdp

can any body help me to over ride the callerid?

Thanks,
Max Alex
Voip Developer
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[asterisk-users] Asterisk custom functions

[Max Alex]

Hi All,
i have centos5 system, i have installed asterisk 1.4 branch.
i havedone realtime connection with odbc to pgsql.
i have created custom functions in func_odbc.conf, all dsn setup and
connection is working fine,
but custom functions are not being registered to asterisk.

i have given queries to functions and using that functions in dialplan.
but it is always gives me function is not registered.

can any body explain how to register custom functions in asterisk?

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Dial issue

[Max Alex]

Hi,
can you please confirm that DTMF is working properly or not?

Thanks,
Max Alex
Voip Developer



On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote:

 Hi, when I make a call I need that the caller can** hang up by dialing ***(H 
 option in Dial command), the call but it don´t work.

 Command

 EXEC DIAL Zap/g1/433391|20|H

 In CLI...
  -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H)
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called g1/433391
 -- Zap/1-1 is ringing
 -- Zap/1-1 answered SIP/510093-082160f0
 (--- At this moment I press * several times, but nothing happens
 Then I hung up the phone--)
 -- Hungup 'Zap/1-1'


 Any Ideas?


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Re: [asterisk-users] Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)

[Max Alex]

Hi Hiren,
Can  you please confirm the php-gd is properly installed?

Thanks,
Max Alex
Voip Developer



On Tue, Sep 9, 2008 at 4:20 PM, Hiren Mistry
[EMAIL PROTECTED]wrote:


 Dear All,

  I have configured here Asterisk-stat (Call Detail Records)for 
 CDR ANALYSER. Here I am facing problem in web analyser when I Selection
 of the day as I require it can get data from asterisk postgres database. But
 in bottom side I have not seen graphical chart and I also can't make
 export file.  CDR (Call Detail Records)
   umber of calls : 21534   * - Call Logs - *  [image: Back to Top]
   *  Calldate
 http://192.168.0.6/asterisk-stat-v2/?s=1t=stitle=atmenu=current_page=0order=calldatesens=ASCposted=1Period=Dayfrommonth=fromstatsmonth=2008-09tomonth=tostatsmonth=2008-09fromday=truefromstatsday_sday=01fromstatsmonth_sday=2008-09today=truetostatsday_sday=01tostatsmonth_sday=2008-09dsttype=1sourcetype=1clidtype=1channel=resulttype=mindst=src=clid=
 *   * Channel *   * Source *   * Clid *   * Lastapp *   * Lastdata *   * Dst
 http://192.168.0.6/asterisk-stat-v2/?s=1t=stitle=atmenu=current_page=0order=dstsens=ASCposted=1Period=Dayfrommonth=fromstatsmonth=2008-09tomonth=tostatsmonth=2008-09fromday=truefromstatsday_sday=01fromstatsmonth_sday=2008-09today=truetostatsday_sday=01tostatsmonth_sday=2008-09dsttype=1sourcetype=1clidtype=1channel=resulttype=mindst=src=clid=
 *   * APP *   * Disposition *   *  Duration
 http://192.168.0.6/asterisk-stat-v2/?s=1t=stitle=atmenu=current_page=0order=durationsens=ASCposted=1Period=Dayfrommonth=fromstatsmonth=2008-09tomonth=tostatsmonth=2008-09fromday=truefromstatsday_sday=01fromstatsmonth_sday=2008-09today=truetostatsday_sday=01tostatsmonth_sday=2008-09dsttype=1sourcetype=1clidtype=1channel=resulttype=mindst=src=clid=
 *   * Userfield *   * Accountcode *   1.  2008-09-01 23:59 Zap/84-...
 992 992 BackGround ivr_menu/mainmenu/PRESS_03 s
  ANSWERED 00:35 Ij
2.  2008-09-01 23:59 Zap/83-... 975 975 WaitExten 10 s
  ANSWERED 00:19 IP
3.  2008-09-01 23:59 Zap/82-... 971 971 BackGround
 ivr_menu/mainmenu/guj_promp s
  ANSWERED 00:34 IGj
4.  2008-09-01 23:59 Zap/81-... 7965 7965 Dial ZAP/R2/501|60 s
  ANSWERED 06:12 ATA
5.  2008-09-01 23:58 Zap/80-... 972 972 WaitExten 10 s
  ANSWERED 00:20 Ij


 *I have not seen below graphical chart and I also can't make export file.*
 * *--* *
 --* *
 --* *
 --
  *TOTAL*

  * **ASTERISK MINUTES*
 Date Duration GRAPHIC
  CALLSACT
 2008-09-01   3
  27789


Export PDF file  Export CSV file

 --
 With Regards,
 Hiren Mistry


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[asterisk-users] Help about the Rxfax on asterisk

[Max Alex]

Hi all,
I have a trixbox2.6.1 on my one server,
i have configured sangoma A200/Remora FXO/FXS Analog AFT card on that
server,
from my zap line the incoming faxes are coming, i have setup the did for zap
channel.
my question is when i am getting any faxes, asterisk shows me rxfax
execution and suddently asterisk crashes
and i can't get email notification for received faxes.

any one help me about the crashes of asterisk?

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Asterisk CDR Problem

[Max Alex]

Hi,
let me know that you have configured properly in res_pgsql.conf in asterisk
with proper, and it is connected properly to database with database details.

Thanks,
Max Alex
Voip Developer



On Fri, Aug 29, 2008 at 10:26 AM, Hiren Mistry 
[EMAIL PROTECTED] wrote:


 Hi ,
 I have check zapte.conf in and after make some correction that problem
 solve.

 But now I am facing other problem. We are using here Postgres Database and
 the data from CLI it can't insert in Postgres Database. I have also here
 mention below cdr_pgsql.conf, modules.conf and cdr.conf

 cdr.conf -- Below
 [general]

 [csv]
 usegmtime=yes ;log date/time in GMT
 loguniqueid=yes ;log uniqueid
 loguserfield=yes ;log user field

 --
 cdr_pgsql.conf -- Below

 [global]
 hostname=localhost
 ;hostname=122.160.10.81
 port=5432
 dbname=asterisk
 password=postgres
 user=postgres
 table=cdr

 --
 modules.conf -- Below

 [modules]
 autoload=yes
 ;preload = res_odbc.so
 ;preload = res_config_odbc.so

 noload = pbx_gtkconsole.so
 ;load = pbx_gtkconsole.so
 noload = pbx_kdeconsole.so
 load = res_musiconhold.so
 noload = chan_alsa.so



 --
 OUTPUT on CLI --- Below
 localhost*CLI cdr status
 CDR logging: enabled
 CDR mode: simple
 CDR registered backend: csv
 CDR registered backend: cdr_manager
 CDR registered backend: cdr-custom
 localhost*CLI

 also when I load manually cdr_pgsql.so on CLI then it show error which is
 also I describe below
 localhost*CLI module load cdr_pgsql.so
 [Aug 29 10:22:21] WARNING[8984]: loader.c:362 load_dynamic_module: Error
 loading module 'cdr_pgsql.so': libpq.so.5: cannot open shared object file:
 No such file or directory
 [Aug 29 10:22:21] WARNING[8984]: loader.c:614 load_resource: Module
 'cdr_pgsql.so' could not be loaded.


 So, Please guide me for load Postgres module in asterisk for CDR Database.

   Subject:
 Re: [asterisk-users] Asterisk CLI Show Error :- (**Unknown**) instead of
 (Zap/22-1, )  From:
 Max Alex [EMAIL PROTECTED] [EMAIL PROTECTED]  Date:
 Thu, 28 Aug 2008 11:48:16 +0530To:
 Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com asterisk-users@lists.digium.com
 Hi Hiren,
 Have you properly configured the zap channels in asterisk,
 which device have you configured in asterisk with zaptel?

 let me know the dial plan for ivr.

 Thanks,
 Max Alex
 Voip Developer



 On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry 
 [EMAIL PROTECTED] wrote:

 Hi, Everybody,

 I am planning to make a new IVR on Asterisk I have Installed zaptel ,
 libpri, asterisk, asterisk-addon on CentOS 5
 I also start service of zaptel and asterisk it start successfully. But
 when goto asterisk CLI prompt and check this IVR then all call string
 with (**Unknown**) instead of (Zap/22-1, ) and I have also 3 other
 Asterisk base IVR which is also on CentOS.
 [Asterisk CLI  Executing [EMAIL PROTECTED]:1] Answer(**Unknown**, ) in
 new stack ]

 Please Help me for Configuring this IVR.

 --
 With Regards,
 Hiren Mistry



 --
 With Regards,
 Hiren Mistry


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Re: [asterisk-users] Asterisk CLI Show Error :- (**Unknown**) instead of (Zap/22-1, )

[Max Alex]

Hi Hiren,
Have you properly configured the zap channels in asterisk,
which device have you configured in asterisk with zaptel?

let me know the dial plan for ivr.

Thanks,
Max Alex
Voip Developer



On Thu, Aug 28, 2008 at 11:40 AM, Hiren Mistry 
[EMAIL PROTECTED] wrote:


 Hi, Everybody,

 I am planning to make a new IVR on Asterisk I have Installed zaptel ,
 libpri, asterisk, asterisk-addon on CentOS 5
 I also start service of zaptel and asterisk it start successfully. But
 when goto asterisk CLI prompt and check this IVR then all call string
 with (**Unknown**) instead of (Zap/22-1, ) and I have also 3 other
 Asterisk base IVR which is also on CentOS.
 [Asterisk CLI  Executing [EMAIL PROTECTED]:1] Answer(**Unknown**, ) in
 new stack ]

 Please Help me for Configuring this IVR.

 --
 With Regards,
 Hiren Mistry


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[asterisk-users] Voicemail has issues with DTMF

[Max Alex]

Hi everybody,
I have linksys phone at my location,
i am using asterisk version 1.4.19,
I have a issue regarding dtmf mode, i have set the Asterisk DTMF mode to
Auto in order to eliminate Asterisk effect on the DTMF transmission. Both
Inband and AVT from Linksys worked with PSTN IVR.
But, We have the issue why Asterisk Voicemail doesn't work with Linksys set
to Inband and Asterisk set to Auto.
And what is the reply of asterisk while the dtmf configuration like this?
Anyone please help me for this issue, i have searched many pages but i
haven't found the exact solution or reason for this?


-- 
Thanks,
Max Alex
Voip Developer
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[asterisk-users] Blind Transfer is not working in incoming calls

[Max Alex]

Hi Everybody,
i have installed asterisk 1.4.19 on my box,
I have setup agi script which is used while incoming and outgoing calls.
It will find the users for incoming and calls to them which is registered in
asterisk,
I have a setup *# for blind transfer to call any outbound or inbound
numbers.
when i am calling any outbound call and the calls are connected with my sip
peer, then i am pressing *# for blind transfer, it will ask me to enter the
transfer number and it is working,
But when an incoming call to my sip user and they are connected the *# is
not worked even the transfer prompt is also played, and dtmf is also set
properly.

But i am not getting why the incoming call is not transfer to any other
number?
Please help for this issue!


-- 
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Dealy while taking

[Max Weltz]

Hello,

if you give more information about your configuration, it will be  
easier to help you.
Are both clients on the same network or are they separated by a NAT?  
What is the configuration of these SIP clients in sip.conf? Do you  
experience these delays with other soft phones, with non-SIP calls?

Regards,

Max


On Jan 11, 2008, at 14:41 , pgck nirukshitha wrote:

 Hi All

I am getting some delay while taking with software phone. I am  
 using Xlite software phone in both side. Please help me to reduce  
 this delay.

 Regards
 Niru




   
 
 Be a better friend, newshound, and
 know-it-all with Yahoo! Mobile.  Try it now.  
 http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ


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Re: [asterisk-users] [asterisk-dev] trunk working under windows!

[Max McGraw]

 Drew Gibson wrote:

 but ... why?

 so  windows  lawyers can sneak a few patents thru the patent office
 and sue Digium for patent infringement.

 I am not criticizing Zoa or Luigi here, just reflecting on what ends up
 happening eventually.

 Think  BSD code into windows, think file  receive a few patents for
 stolen ideas, think sue linux  open source for patent infringement.

 In my humble opinion.

 I applaud the technical merit of the effort to port things to windows,
 but please remember that you are aiding and abetting the enemy.

 Obviously, it it your time and your dime to do with as you please...
 but you may end up biting the hand that feeds you.


  Zoa wrote:
 
  Cool, i'll help out a bit with the windows port,  i will start right
  away with a new project on asteriskguru making nightly executable builds
  and installers - will post the links in -users when i'm done.
 
  Well done luigi, this will make it a lot easier for a lot of non linux
  guys to make their first steps in the asterisk world
 
  Crossposted to -users.
 
  Zoa
 
  Luigi Rizzo wrote:
 
 
  As a result of the commit below, now trunk can be built and run under
  Windows/cygwin, including the building of modules.
 
 
==

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[asterisk-users] Extensions Configuration

[Max Clark]

Hi all,

I am building out a new platform and I need help with a couple of
items. I need to have an extension 101 that is public (on business
cards, in the directory, etc...) however I want this extension to
exist as a hunt group with a ring all strategy so two phones (107
which is the private extension for the 101 user is run, and the 102
extension). The 107 extension should not have a separate voicemail and
when the user at 107 presses the messages button they need to log into
the 101 mailbox. When 107 dials other users internally it should show
the callerid as 101.

What is the best way to configure asterisk to to this?

Second question, for the hunt groups I want to change the callerid
display for incoming calls so the phone displays Boss's
Line:123456789, but I want to make sure that when the user redials
via the phone directory the number 123456789 is dialed directly. How
do I change the caller id display for inbound calls and still have the
directory work properly?

Thanks in advance,
Max

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[asterisk-users] Random unknown codec format IAX calls

[Max Ochoa]

   I seem to be having a problem that I have narrowed down to a 
disagreement on codec negotiation or codec setup of some kind in an IAX 
peering arrangement. Here's a non-ASCII art version of the setup:


DID origination provider
 via SIP/gsm
 to
Call routing asterisk server
 via IAX/gsm
 to
Client asterisk server
 via SIP/ulaw
 to
Polycom 501 UA

The problem that occurs sporadically (1/10 times) is the call will 
complete and stay active, but there is no audio. There is a channel open 
all the way to the phone, and the codec (gsm) is shown as the format for 
the call for the SIP channel and the IAX channel from the Call routing 
server to the Client asterisk server. However, the Client asterisk 
server shows the call format as unknown when a call is open that has 
no audio. The codec was originally forced to to gsm, then forced to 
ulaw, then set for any (allow=all) with the same results. Here's the 
output from the console on calls with these symptoms. IAX debug output 
looked the same for calls that had audio and those that did not, so I'll 
spare posting that.


Asterisk console (verbose):
===
 -- Accepting AUTHENTICATED call from 10.3.0.1 http://10.3.0.1:
   requested format = gsm,
   requested prefs = (),
   actual format = gsm,
   host prefs = (ulaw|alaw|gsm),
   priority = mine
   -- Executing Wait(IAX2/customer-8, 0) in new stack
   -- Executing Set(IAX2/customer-8, _CONTEXTNAME=customer) in new 
stack

   -- Executing Set(IAX2/customer-8, _VMEXTEN=100) in new stack
   -- Executing Set(IAX2/customer-8, 
_VOIP_SERVER=customer.voip.domain.net 
http://brasovan.voip.bestserversllc.net) in new stack

   -- Executing Set(IAX2/customer-8, TIMEOUT(digit)=5) in new stack
   -- Digit timeout set to 5
   -- Executing Set(IAX2/customer-8, TIMEOUT(response)=6) in new stack
   -- Response timeout set to 6
   -- Executing Dial(IAX2/customer-8, SIP/100-customerSIP/101
-customer|25|tr) in new stack
   -- Called 100-customer
   -- Called 101-customer
   -- SIP/100-customer-081940e0 is ringing
   -- SIP/101-customer-081a1c70 is ringing
   -- SIP/100-customer-081940e0 answered IAX2/customer-8

customer*CLI iax2 show channels
Channel   Peer UsernameID (Lo/Rem)  Seq 
(Tx/Rx)  Lag  Jitter  JitBuf  Format
IAX2/customer-8   10.3.0.1 http://10.3.0.1 customer
8/3  00014/00010  00079ms  -0001ms  ms  unknow

1 active IAX channel

customer*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold 
Last Message   
10.0.0.103 http://10.0.0.103   100-custo  54eb074262d  
00102/0  ulaw  No   Tx: ACK
1 active SIP channel

===

There is a vtun IP tunnel between the Call routing asterisk server and 
the Client asterisk server (the 10.3.0.0/24 subnet.) The 10.0.0.0/24 
subnet is the client's LAN.


Any tips / ideas on what to try next are appreciated.

- Max

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[asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved

[Max Bergmann]

Max Bergmann schrieb:



How can i programming a Cisco 7961 to be used as busy lamp field?

my configs :

sccp.conf :

[devices]
type= 7961
tzoffset= 0
autologin   = 601
speeddial   = *31, Hanna  -- other SIP telefon

extensions.conf :

exten = *31,hint,SIP/hanna
exten = *34,hint,SCCP/601


on SIP Telefon ( SNOM 360 ) everything functions good and i have busy 
lamp when cisco telefon Offhook, but differently does not function

any idea ?

Any input is greatly appreciated.


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 I have solved my problem, thank you for your help


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[asterisk-users] cisco 7961 , asterisk and busy lamp

[Max Bergmann]

How can i programming a Cisco 7961 to be used as busy lamp field?

my configs :

sccp.conf :

[devices]
type= 7961
tzoffset= 0
autologin   = 601
speeddial   = *31, Hanna  -- other SIP telefon

extensions.conf :

exten = *31,hint,SIP/hanna
exten = *34,hint,SCCP/601


on SIP Telefon ( SNOM 360 ) everything functions good and i have busy 
lamp when cisco telefon Offhook, but differently does not function

any idea ?

Any input is greatly appreciated.


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[asterisk-users] Asterisk 1.4 Schedule and Features/Changes

[Max Clark]

Hi all,

Asterisk 1.4 was originally scheduled to be released early July
2006. Is there an update on the expected release of this version?
Also is there a changelog or feature list available that lists the
differences over 1.2?

TIA,
Max

--
Max Clark
http://www.clarksys.com
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[Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 1/3

[Max Glucksmann]

Hello Friends,



I was looking for help to set AstCC with different sound
files for the Spanish syntax by setting the language to the sound folder name.
It worked all right except that the syntax used was English. It just didnt
sound right.



I couldnt find much support to compile a new say.c
module for Asterisk, but I believe it is better that way. This application may
not be of the common interest, but many people may want to have support for
different sound files in a different syntax than English, and in the same
Asterisk server; so, if anyone is interested, the code is posted in the two
following e-mails. Use the function mysaynumber instead of
calling the say_number AGI function.



Max Glucksmann

e-mail: [EMAIL PROTECTED]

Web: http://www.comtel-networks.com



USA

Phone: 1 (877) 467-2877
 ext. 1011001

Fax: (954) 827-0990



Venezuela

Teléfono: (0500) MAXITEL
 ext. 1011001

Fax: (0212) 953-0769






BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
END:VCARD
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[Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 2/3

[Max Glucksmann]

Hello,



As posted on the first of these three e-mails, heres
the function that supports multiple sound folders for the Spanish syntax.



Be sure to store in $sound_map the folder name, which must
also be set as the language.



sub mysaynumber() {

    my ($number) = @_;

    my $res;

##

### Support for separate sound file in spanish syntax:

##

    if ( index( $sound_map, \_es )  -1 )
{

    my $thousands = int ( $number / 1000 );

    my $remains = ( $number - $thousands * 1000
);

    $AGI-verbose( Spanish Syntax for:
$sound_map., $verbose );

    $AGI-verbose( Number:
$number., $verbose );

    if ( $thousands  0  $thousands
 30 ) {

    if ( $thousands  1 ) {

    $res = mystreamfile(digits/$sound_map/$thousands);

    $AGI-verbose(
Thousands: $thousands, $verbose );

    }

    $res =
mystreamfile(digits/$sound_map/mil);

    $AGI-verbose( Thousand
Sound: digits/$sound_map/mil, $verbose );

    }

    my $hundreds = int ( $remains / 100 );

    $remains = $remains - $hundreds * 100;

    if ( $hundreds ) {

    if ( $hundreds  1 || ( $hundreds
== 1  $remains == 0 ) ) {

    $hundreds = $hundreds * 100;

    $res =
mystreamfile(digits/$sound_map/$hundreds);

    $AGI-verbose(
Hundreds: $hundreds, $verbose );

    } else {

    $res =
mystreamfile(digits/$sound_map/100-and);

    $AGI-verbose(
Hundred and sound file: 100-and, $verbose );

    }

    }



The rest of the function continues on the third e-mail



With best regards,



Max Glucksmann

e-mail: [EMAIL PROTECTED]

Web: http://www.comtel-networks.com



USA

Phone: 1 (877) 467-2877
 ext. 1011001

Fax: (954) 827-0990



Venezuela

Teléfono: (0500) MAXITEL
 ext. 1011001

Fax: (0212) 953-0769






BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
END:VCARD
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[Asterisk-Users] Multiple Sound Files Folders - Spanish Syntax. AstCC AGI. 3/3

[Max Glucksmann]

The function continues here from e-mail no. 2:



    my $decades = int ( $remains / 10 );

    $remains = $remains - $decades * 10;

    if ( $decades  3 ) {

    $decades = $decades * 10;

    $res =
mystreamfile(digits/$sound_map/$decades);

    $AGI-verbose(
High Decades: $decades, $verbose );

    if ( $remains 
1 ) {

    $res =
mystreamfile(digits/$sound_map/and);

    $res =
mystreamfile(digits/$sound_map/$remains);

    $AGI-verbose( With
$remains, $verbose );

    } elsif ( $remains == 1 ) {

    $res =
mystreamfile(digits/$sound_map/and);

    $res =
mystreamfile(digits/$sound_map/1M);

    $AGI-verbose( With
1M, $verbose );

    }

    } else {

    $decades = $decades * 10 + $remains;

    $res =
mystreamfile(digits/$sound_map/$decades);

    $AGI-verbose(
Low Decades: $decades, $verbose );

    }

##



    } else {

    $AGI-verbose( Playing Number
without Custom Syntax: $number, $verbose );

    $res = $AGI-say_number($number,
0123456789);

    $res =  if $res eq
0;

    }

    $AGI-verbose( RES: $res, $verbose )
if ( $config{debug_agi} eq YES);

    $res = sprintf(%c, $res) if ( length(
$res ) );

    return $res;

}



Hope this helps someone as it worked for me.



With best regards,



Max Glucksmann

e-mail: [EMAIL PROTECTED]

Web: http://www.comtel-networks.com



USA

Phone: 1 (877) 467-2877
 ext. 1011001

Fax: (954) 827-0990



Venezuela

Teléfono: (0500) MAXITEL
 ext. 1011001

Fax: (0212) 953-0769






BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
END:VCARD
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[Asterisk-Users] Multiple Sound Folder Support for Same Language Syntax

[Max Glucksmann]

Hello folks,

I’m setting up in one server two AstCC IVRs with different sound files but in 
the same language: Spanish. As you may already know, Spanish differs slightly 
in syntax from English. I tried setting the language to the folder names and 
sounds are taken correctly, but the syntax used is still english... 

I took a look in say.c and found the syntax section. It seems easy to patch so 
it supports this new scheme that uses multiple folders using the same syntax 
and different files; something like appending _es to the folder names so the 
module recognizes it is Spanish would do it for my case, but I can’t get to 
compile correctly the file.

In any case, this is the function that I would modify to include the new 
folders’ names:
static int ast_say_number_full_es

To be something like:
static int ast_say_number_full_new_folder_1_es
static int ast_say_number_full_new_folder_2_es

That would be a quick and good solution, considering that I won’t be adding 
IVRs everyday.

To make something better, I could write a function that called the original for 
every directory that ends in _es, or more likely to support every language 
searching for the suffix.
 
Could someone please guide me to recompile after making the modifications?

I’d be happy to publish whatever I come up with; it doesn’t really seem to be 
too complicated but it has been a very long time since I compiled my last C 
program ☺

Your help will be greatly appreciated.

With best regards,

Max Glucksmann
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
 
Venezuela
Teléfono: (0500) MAXITEL – ext. 1011001
Fax: (0212) 953-0769
 
USA
Phone: 1 (877) 467-2877 – ext. 1011001
Fax: (954) 827-0990

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[Asterisk-Users] Multiple Sound Folders Support for Same Language (Syntax)

[Max Glucksmann]

Hello folks,



Im setting up in one server two AstCC IVRs with
different sound files but in the same language: Spanish. As you may already
know, Spanish differs slightly in syntax from English. I tried setting the
language to the folder names and sounds are taken correctly, but the syntax
used is still english... 



I took a look in say.c and found the syntax section. It
seems easy to patch so it supports this new scheme that uses multiple folders
using the same syntax and different files; something like appending _es to the
folder names so the module recognizes it is Spanish would do it for my case,
but I cant get to compile correctly the file.



In any case, this is the function that I would modify to
include the new folders names:

static int ast_say_number_full_es



To be something like:

static int ast_say_number_full_new_folder_1_es

static int ast_say_number_full_new_folder_2_es



That would be a quick and good solution, considering that I
wont be adding IVRs everyday.



To make something better, I could write a function that
called the original for every directory that ends in _es, or more likely to
support every language searching for the suffix.

 

Could someone please guide me to recompile after making the
modifications?



Id be happy to publish whatever I come up with; it
doesnt really seem to be too complicated but it has been a very long
time since I compiled my last C program J



Your help will be greatly appreciated.



With best regards,



Max Glucksmann

e-mail: [EMAIL PROTECTED]

Web: http://www.comtel-networks.com



Venezuela

Teléfono: (0500) MAXITEL
 ext. 1011001

Fax: (0212) 953-0769



USA

Phone: 1 (877) 467-2877  ext. 1011001

Fax: (954) 827-0990





Comtel Networks, Corp. - Proprietary and
Confidential








BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
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[Asterisk-Users] Realtime Content on LCD Display

[Max Glucksmann]

Hello,

Anyone knows a way to show real-time content from a DB into the LCD display
of an IP phone, like any 79xx?

If someone knows which phone is capable of doing and how, like using XML
files, please advise.

Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
 
Venezuela
Teléfono: (0500) MAXITEL – ext. 1011001
Fax: (0212) 953-0769
 
USA
Phone: 1 (877) 467-2877 – ext. 1011001
Fax: (954) 671-6800
BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
END:VCARD
BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
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[Asterisk-Users] RE: [Asterisk-Users ] RE: Monitor a call in progress. (Steve Totaro)

[Max Glucksmann]

Steve,

You wrote this referring to monitoring a call in Asterisk, how about from an
IP phones LCD display screen:

1.  go to www.google.com
2.  type asterisk monitor application
3.  click on the first result
4.  read and implement
5.  google is your friend 

I hope I made myself clear too ;-P

Moreover, which phone can we use? We have a call shop cashier attended
feature for call shops, but still need to display the call to the booth
user...

Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
 
Venezuela
Teléfono: (0500) MAXITEL – ext. 1011001
Fax: (0212) 953-0769
 
USA
Phone: 1 (877) 467-2877 – ext. 1011001
Fax: (954) 671-6800
BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
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Re: [Asterisk-Users] Asterisk Follow Me

[Max Clark]

Thank You.

On 2/21/06, C F [EMAIL PROTECTED] wrote:
 http://bugs.digium.com/view.php?id=5574
 That is a patch that will do just that.

 On 2/21/06, Max Clark [EMAIL PROTECTED] wrote:
  Hi all,
 
  I am interested in a follow me script for Asterisk - specifically I am
  looking for one that will prompt the calling party to record their
  name and then call through a list of numbers playing the recording. If
  a digit is pressed by the recipient then the call is put through.
 
  Is there anything like this available as an example for Asterisk?
 
  TIA,
  Max
 
  --
  Max Clark
  http://www.clarksys.com
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 Max Clark
 http://www.clarksys.com
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[Asterisk-Users] Fromuser required but overrides SetCallerID

[Max Clark]

Hi all,

I have an asterisk box connecting to a SER instance for outbound
(termination) calling. In order to authenticate with SER it seems that
I have to use fromuser in the sip.conf in the peer section for the
SER connection - with fromuser set I can make calls, without it I get
a Forbidden - wrong password on authentication for INVITE error.

The problem is that setting fromuser in the sip.conf overrides
anything that I have set in the dialplan with SetCallerID. How do I
work around this?

TIA,
Max

--
Max Clark
http://www.clarksys.com
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[Asterisk-Users] DTMF Tones in RTP Payload as Well as in Events = Duplicate Tones

[Max Glucksmann]

Dear friends,

As I commented some while ago in the list, occasionally when DTMF Tones are
sent, they appear in RTP Payload and in Events too, producing duplicate
tones being recognized. This behavior happens in Asterisk as well as in
Gateways such as Cisco, for which we had the opportunity to observe the
error and extensively debug it.
 
We ended up recognizing good digits by adjusting audio gain in the Cisco
IOS, but now some calls' volume is just too low to hear comfortably.

If you could let me know how to adjust reception gain in * it would help us
treat the problem from a different angle.

Resuming, we need to find support to modify rtp.c or dsp.c in order to
silence audio when tones are sent (received in *) from the user to * through
providers using CODECS G.723 and G.721 and DTMF recognition method RFC2833.

Regards,
Max Glucksmann
e-mail: [EMAIL PROTECTED]
Web: http://www.comtel-networks.com
BEGIN:VCARD
VERSION:2.1
N:Glucksmann;Max
FN:Max Glucksmann (Fax del trabajo)
ORG:ComTel Networks, Corp.
TITLE:Director
TEL;WORK;VOICE:+1 (877) 467-2877
TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835)
TEL;CELL;VOICE:+58 (414) 250-0909
TEL;WORK;FAX:+1 (954) 671-6800
TEL;HOME;FAX:+58 (212) 285-3320
ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de=
 Am=E9rica
EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800
REV:20051212T222729Z
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[Asterisk-Users] Asterisk Follow Me

[Max Clark]

Hi all,

I am interested in a follow me script for Asterisk - specifically I am
looking for one that will prompt the calling party to record their
name and then call through a list of numbers playing the recording. If
a digit is pressed by the recipient then the call is put through.

Is there anything like this available as an example for Asterisk?

TIA,
Max

--
Max Clark
http://www.clarksys.com
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Re: [Asterisk-Users] Teliax Down?

[Max Clark]

I hate to burst your bubble but DOS attacks are a fact of life for IP based services. The bigger you get the more of a target you are. There are a ton of DOS prevention/mitigation appliances/services available in today's world. But they all rely on the same thing: having more bandwidth/capacity than your attacker.


I've seen DOS attacks against ISP customers of mine that were pushing over a million packets per second across 50+ peering points. Not many networks can absorb that kind of thing.

If your phones are that critical to your business you need to get dedicated service (aka T1), or switch to a service with static registration that can be protected with a good firewall.

Max
On 1/23/06, JCC [EMAIL PROTECTED] wrote:


I've had problems for the last couple of weeks regarding incoming calls. Cant hear the party calling me (their voice sounds garbled/scrambled). If you haven't done so yet, I would recommend you post your complaint on their online forum as well under 'bugs'. You usually get some good responses from other Teliax users regarding the problem.






From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Ross CSent: Friday, January 20, 2006 8:40 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Teliax Down?

I was having trouble too. I had trouble yesterday as well. I called and David said it was a "massive DDOS". Seems to get fixed pretty quickly when it does happen (5 minutes or so); however, for a business, 5 minutes without phones (people can't get a hold of your company) isn't really acceptable IMO.


Also on co3. I couldn't even access their website during that time…





From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Rusty DekemaSent: Friday, January 20, 2006 5:42 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Teliax Down?

Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; 
voip-co3.teliax.com. -Rusty___
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