Re: [asterisk-users] what can we do with lost voice packet on a congestioned VPN?

[Julio Arruda]

nik600 wrote:
> Hi to all
> in a scenario where:
> 
> - the bandwith is shared with other traffic (HTTP,VPN,ecc)
> - the PBX is on a remote VPN peer
> - due to many reasons Qos is not usable
> 
> There is a IAX trunk between 2 Asterisk 1.4 i've tried different
> codecs (ulaw,alaw,gsm) but the main problem still remain the same: too
> many voice packet get lost.
> 
> The main problem is surely on the network, but the strange thing is
> that on the same network there is an H323 trunk from an Alcatel and a
> Cisco CCM (using g711 codec) and in that case the voice isn't so bad!
> 
> i've tried to enable jitterbuffer but i can't notice some difference.
> 
I think if you have packet loss, a jitterbuffer would make no difference..
SOmething like PLC (packet loss concealment maybe ?) come to mind ?

Have you tried ILBC ? or SPEEX ? G.711 should be resilient, but I 
understand iLBC and Speex both have some tricks to handle PLC.
Also, GIPS had some kind of magic sauce they offered (code-wise) to 
offer PLC to vendors ??

> Is there something else that i can do?
> 
> Thanks to all
> 


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Re: [asterisk-users] MagicJack and Skype call quality

[Julio Arruda]

Jason Aarons (US) wrote:
> My understanding is Skype's secret is using the iLBC codec, which Cisco
> has also licensed for their 79X2 models as well.  I travel and lot and
> in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator
> will fail the Skype client will work.  The iLBC codec can really handle
> packet loss.

AFAIK, not iLBC, but another GIPS codec.
ILBC is present in some grandstream phones from what I remember, not in 
the Cisco 7912 as one example, not sure about other phones.
In a word with still many PSTN gateways out there that don't support it, 
ILBC in the SIP UA side only can help that much :-)..

> 
> Skype High Quality Video with the Logitech Orbit AF on both ends is
> awesome. I got my family a set for Fathers day. Just amazing video
> quality. Uses a On2 VP-7 codec that has much lower cpu and other
> benefits over h.264.
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
> Underwood
> Sent: Saturday, July 12, 2008 3:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] MagicJack quality
> 
> Tzafrir Cohen wrote:
>> On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote:
>>   
>>> C. Savinovich wrote:
>>> 
 I am puzzled by the quality of magicjack.  I keep trying to figure
> out how
 they can the quality be that adequate.  Since Skype also has an
> excellent
 quality, that leaves me to believe that software based calls
> (softphones)
 could have and advantage over hardphones, provided there is a
> parameter that
 those 2 companies are addressing.

 Anyone's thoughts on this?

 CS
   
   
>>> I don't know what Magic-jack does (I've never actually seen one), but
> I 
>>> know the key thing about Skype that impresses people - its wideband 
>>> voice codec. A lot of people poo-poo the idea that wideband voice has
> 
>>> value in a phone call. They are either close to deaf, or have never 
>>> tried it. Clarity is profoundly improved. Skype seems to use various 
>>> tricks to keep the packet flow smooth, but its wideband that makes it
> 
>>> sound better than the PSTN.
>>>
>>> You might think a standard phone plugged into an adaptor, like a 
>>> Magic-jack, would be limited to narrow band voice, as that is all the
> 
>>> phone was designed for. It turns out most phones only aggressively 
>>> filter at the low end of the band. They let a lot of energy above
> 4kHz 
>>> through, and they do generally sound better through a wideband codec.
>>>
>>> Many modern line interface chips are actually capable of running in a
> 
>>> 16k samples/second mode, even though most are programmed for 8k 
>>> samples/second. I think the ones on the TDM400P type cards can. Some 
>>> from Silicon Labs certainly can, and chips from Zarlink and others
> can.
>>> 
>> The DAA in those cards can work in 16Hz. So they can send higher
> quality
>> samples to the telco. Provided Zaptel supports it. But then again, it
>> will get lost as soon as it gets converted to digital at the telco,
>> right?
>>   
> I guess I wasn't clear. What I said was only useful for a SLIC to phone 
> connection. It won't be of any benefit for a DAA to PSTN exchange 
> connection, for the reason you state.
>> Anyway, the ProSLIC chip does not seem to support it. 
>>   
> Silicon Labs make a Wideband ProSLIC, Si 3216, which is, er, wideband. 
> As I said before, Zarlink and other make them too.
> 
> Regards,
> Steve


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Re: [asterisk-users] Sharing unused minutes between Asterisk users

[Julio Arruda]

Klaverstyn, David C wrote:
> Hi All,
> 
>  
> 
> I was under the impression that I found a WEB site about two years or so
> ago that allowed Asterisk users to place free calls between each other
> that used up users un-used minutes/calls.  I though the site was IAXtel
> but that does not seem to be the case.
> 
>  
> 
> As an example I have a plan with a VSP.  They allow a certain number of
> calls every month but I only use 20% of the allocation.  I was wanting
> to let other people around the world to utilise the additional calls I
> have.  Is there such a WEB site that allows us to connect our Asterisk
> servers together to utilise the otherwise unused calls? 

I would assume this would be against the Terms&Conditions/AUP of a VOIP 
provider..
But, FWD out had something in these lines, where you would 'offer' your 
line for outbound, it was supposed to be related to the PSTN line from 
what I remember.

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Re: [asterisk-users] Asterisk H.248 Support

[Julio Arruda]

Chad Whitten wrote:
> Im not looking for a call agent but rather gateway functionality for a
> lab setting to provide dial tone v h.248 to client gateways.
> 

 From what I understand, the Media Gateway (in Megaco world language), 
is the 'client' if you will, while the call server/softswitch/call 
agent, is the 'inteligence'.
If you have a gateway (either a small one or a trunk one), you would 
need yes, a Call Agent.
I found this interesting link that has a summary on this..
http://www.javvin.com/protocolVOIP.html
Who provides the Dial Tone would be, I assume, the gateway, the 
"client", either a "line side" one, like the Calix, MG9000 or others, or 
a trunk side gateway, like the MG15K in Nortel NGN deployments.
In a Nortel CS2000/CS2Kc NGN, the 'megaco/h248 call agent' would be the 
call server itself (actually, the gateway controllers, that translate 
from PPVM to H.248, from what I can remember).

You may also want to check if your gateway can't be changed to use 
another protocol, since H.248, AFAIK, is not exactly much supported in 
the OSS world


> On Feb 16, 2008 5:05 PM, Julio Arruda <[EMAIL PROTECTED]> wrote:
>> Chad Whitten wrote:
>>> I have been searching for some documentation that would indicate if
>>> Asterisk supports H.248 and everything I have come across seems to
>>> indicate I should use MGCP which I would agree is a better choice but
>>> unfortunately the equipment I am trying to integrate only does H.248.
>>>
>>> Could anyone point me to something related to this.
>>>
>>
>> I've not seem anything on Asterisk being used as a Call Agent for H.248
>> gateways, but I've seem the question pop-up at least a couple of times
>> in the past.
>> I understand there are quite few gateways out there that support H.248
>> in one flavor or another. I remember Nortel (used to work there) has at
>> least the PVGs/MG7k/15k (high density trunking gateways, quite
>> interesting in terms of carrier grade features, like
>> non-service-interrupting sw upgrades and etc) and some other gear that
>> would run with H.248, also Calix had at least some sw version that would
>> do H.248. Isn't H.248/Megaco a kind of 'son-of-MGCP' ?
>> I wonder how much effort would be required to implement MGC/call-agent
>> capabilities in asterisk..

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Re: [asterisk-users] Asterisk H.248 Support

[Julio Arruda]

Chad Whitten wrote:
> I have been searching for some documentation that would indicate if
> Asterisk supports H.248 and everything I have come across seems to
> indicate I should use MGCP which I would agree is a better choice but
> unfortunately the equipment I am trying to integrate only does H.248.
> 
> Could anyone point me to something related to this.
> 


I've not seem anything on Asterisk being used as a Call Agent for H.248 
gateways, but I've seem the question pop-up at least a couple of times 
in the past.
I understand there are quite few gateways out there that support H.248 
in one flavor or another. I remember Nortel (used to work there) has at 
least the PVGs/MG7k/15k (high density trunking gateways, quite 
interesting in terms of carrier grade features, like 
non-service-interrupting sw upgrades and etc) and some other gear that 
would run with H.248, also Calix had at least some sw version that would 
do H.248. Isn't H.248/Megaco a kind of 'son-of-MGCP' ?
I wonder how much effort would be required to implement MGC/call-agent 
capabilities in asterisk..



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Re: [asterisk-users] switch QOS requirements

[Julio Arruda]

Al lists wrote:
> Theoretically, setting TOS value ( these days called DSCP) wont change
> anything in switch behavior, unless you are using Layer 3 switches.
> What makes a difference in a switch is COS bits, and i'm not sure how
> asterisk sets that.
> I guess to be safe, you would need to create 2 VLANS and in the switch
> define on VLAN  as a high priority VLAN.

At least for quite few years (more than 5), layer 2 switches from Nortel 
(disclaimer, I used to work for NT), would be able to match DSCP (or 
remark DSCP also, based in l3/l4 information) and give priority as 
defined by the user, to specific DSCP #, like EF to highest priority and 
goes on.
I've no reason to believe other vendors don't have at least this capability.

> On Feb 3, 2008 7:06 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> 
>> On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote:
>>
>>> "John Williams" <[EMAIL PROTECTED]> writes:
>>>
 We are tearing out legacy PBX and replacing with Asterisk PBX and new
 LAN for our 90+ person operation.   Question:  what QOS capabilities
 (protocols, etc) does Asterisk support/require in a LAN switch to
>> deliver
 business grade phone service?  Thanks
>>> If you have one switch for the whole network, you're generally fine
>>> without QoS. Switches these days can handle full bandwidth on all
>>> ports at the same time.
>>>
>>> Anyway, Asterisk is no different from other PBX's when it comes to
>>> QoS. Should it turn out that you actually need it on the LAN, just be
>>> sure you set the tos parameters in sip.conf to something that is
>>> prioritized by the switch.
>> It tends to be more of an issue when you're sending calls over a link
>> with limited bandwidth. Usually more of a concern in the router.
>>
>> Michael
>> --
>> Michael Graves
>> mgravesmstvp.com
>> blog.mgraves.org
>> o713-861-4005
>> c713-201-1262
>> sip:[EMAIL PROTECTED]
>> skype mjgraves
>> fwd 54245

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Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

[Julio Arruda]

Fernando Berretta wrote:
> Dear Mindaugas,
> 
> Thanks for your promt response
> 
> I've already tried this but.. it's not working,, what file do you think 
> I should use ? any other idea ?


Fernando,
I've used the official/legal G729 codec sold at www.digium.com in Athlon 
boxes w/ asterisk 1.4 without problems, have you tried this option ?


> Mindaugas Kezys wrote:
>>
>> Rename to codec_g729.so 
>>  
>>
>>
>> Copy to /usr/lib/asterisk/modules
>>
>> chmod 777 codec_g729.so
>>
>>  
>>
>> restart Asterisk
>>
>> show translations
>>
>>  
>>
>> Mindaugas Kezys
>>
>> http://www.kolmisoft.com
>>
>> Advanced Billing for Asterisk PBX
>>
>>  
>>
>> *From:* [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] *On Behalf Of 
>> *Fernando Berretta
>> *Sent:* Monday, November 26, 2007 6:01 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core 
>> processor 4000 + CENTOS 5 + Asterisk 1.4
>>
>>  
>>
>> Dear Mindaugas,
>>
>> I've already download the folowing files for testing
>>
>> codec_g729-ast14-gcc4-glibc-athlon-sse.so 
>>  
>>
>> codec_g729-ast14-gcc4-glibc-core2.so 
>> 
>> codec_g729-ast14-icc-glibc-x86_64-core2.so 
>>  
>>
>>
>> But... no one of them seems to be working


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Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup

[Julio Arruda]

Olivier wrote:
> Hi,
> 
> 1. Is your WiFi network dedicated to VoIP or shared with data applications ?
> How was it designed ?
> For people using WiFi with a laptop, you propably don't need to have dense
> WiFi cells as moving from one cell should be scarce.
> With hand phones, those cells should overlap as it becomes very likely users
> would to move from one location to another while on the phone.
> 
> 2. From this list, the  WiFi hardphones which got only positive answers
> where Siemens Gigaset SL75 and Nokia EXX Series.

Just a side note, a friend of mine working at Aruba (in Latin America) 
in fact does their demos with VOIP using the Nokia EXX, from his 
feedback, customers do like the combo :-)

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Re: [asterisk-users] My G729 problem re-visited

[Julio Arruda]

How many licenses you have (show g729 should give you this info)


Scott Moseman wrote:
> Gateway sends Asterisk an INVITE (using g729)
> Asterisk sends Phone an INVITE (using g711 or g729)
> Phone sends Asterisk an OK (using g711)
> Asterisk sends Gateway an OK (with no RTP choice)
> Gateways ends the conversation
> 
> I can setup the Phone to use g729 and it will reply with an OK for
> g729, but the OK to the Gateway will still stay empty.  Only when I
> enable g711 on the Gateway will this work.  I have experienced this on
> 2 different models of gateways so far.
> 
> I included my config for both the Gateway and the Phone in my original
> message, hoping that maybe I was configuring the Gateway wrong in
> Asterisk?  But no one has said anything so I'm assuming its okay.
> 
> Phone (g729) to Phone (g729) works
> Phone (anything) to Gateway (g711) works
> Phone (anything) to Gateway (g729) does NOT work
> 
> I'm licensed for g729 (although I'm told I should not need it for pass
> through).  And it will transcode when the phone is g729 and the
> gateway is g711.  But for whatever reason I can't use g729 on the
> gateway side of the calling process?
> 
> Thanks,
> Scott
> 
> 
> 
> On 10/12/07, Power, Paul C. <[EMAIL PROTECTED]> wrote:
>> Is the call being dropped or is Asterisk takng a core dump?
>>
>> I have core dump issues with g729 and asterisk 1.4.11, but my set up is
>> a little different than yours...
>>
>>
>>> -Original Message-
>>> From: Scott Moseman
>>> Sent: Friday, October 12, 2007 10:22 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] My G729 problem re-visited
>>>
>>> No ideas on this one from anyone?  I suppose I'm going to
>>> need to pay for some Digium support because this is a really
>>> unusual problem.
>>> Does anyone else have a gateway that speaks g729 to Asterisk
>>> and works?  For whatever reason, Asterisk refuses to reply
>>> back to any of my gateways using g729.  Phone (g729) to phone
>>> (g729) works.  Phone
>>> (g729) to Asterisk to gateway (g711) works.  But attempt g729
>>> between Asterisk and a gateway and it fails -- every time.
>>> Asterisk responds to the gateway but never includes any
>>> codecs in the packet, unless it's g711.  My configurations
>>> are shown below.
>>>
>>> Thanks,
>>> Scott
>>>
>>>
>>> On 9/26/07, Scott Moseman <[EMAIL PROTECTED]> wrote:
 Ok, I built a test system to duplicate my problem and
>>> provide myself a
 platform that I can mess around with to try and break any features.
 My problem is G729 pass-through from a gateway to a phone.
>>> I think I
 even have transcoding working, which makes me more confused
>>> on what's
 wrong with my pass-through. It must be a configuration issue.

 The basics...

 *CLI> core show version
 Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux

 *CLI> show modules like 723
 Module Description Use Count
 codec_g723.so G.723.1 Coder/Decoder 0
 format_g723.so G.723.1 Simple Timestamp File Format 0

 *CLI> show modules like 729
 Module Description Use Count
 codec_g729.so G.729 Coder/Decoder 0
 format_g729.so Raw G729 data 0

 *CLI> show translation
 [truncated]
 g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex
>>> ilbc g726 g722
 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 -
 g729 5 2 2 2 2 2 1 3 - - 11 2 -

 The configuration...

 [gateway]
 type=friend
 host=gateway
 context=default-inbound
 disallow=all
 allow=g729

 [phone]
 type=friend
 context=sip
 host=dynamic
 username=phone
 secret=scott
 dtmfmode=RFC2833
 disallow=all
 allow=g729
 callerid=Scott
 qualify=yes
 canreinvite=no

 exten => 1266,1,Dial(SIP/[number],30,t) exten => 1266,2,Congestion

 exten => 1266,1,Dial(SIP/[number],30)
 exten => 1266,2,Congestion

 (The same results using both of the above dialplans...)

 The environment...

 PSTN -> Gateway -> Asterisk -> Phone

 What I'm seeing works...

 With the gateway setup to send both G711 and G729, it sends
>>> an INVITE
 which includes both G711 and G729 codecs. Asterisk sends an
>>> INVITE to
 my phone with only G729. The call is made and there's a
>>> conversation
 in G711 with the gateway and G729 with the phone. I assume
>>> this means
 Asterisk is transcoding.

 What I"m seeing fails...

 With the gateway setup to send only G729, it sends an INVITE to
 Asterisk which includes only G729. Asterisk send an INVITE to the
 phone using G729, too. The 200 OK from the phone to the Asterisk
 includes G729. The 200 OK going from Asterisk to the
>>> gateway doesn't
 include ANY codec. The call is dropped the moment I pickup
>>> the phone
 to answer the call.

 My question...

 Why does Asterisk

Re: [asterisk-users] G.722: ast_channel_make_compatible failure

[Julio Arruda]

Tzafrir Cohen wrote:
> On Fri, Oct 05, 2007 at 08:12:34AM -0500, Brian West wrote:
>> You can hear and understand someone much better with g722... more  
>> emotion is transfered over the phone when using g722.
>>
>> G722 is free and in the clear. G722.1 and G722.2 are not.
> 
> But speex *Is* free. Including wideband.
> 

How many hardware vendors do support Speex ? From small gateways to big 
trunking gateways ?
I suspect Wideband is one amazing value-proposition for VOIP, in 
countries where bandwidth savings is not the problem to be solved anymore.

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Re: [asterisk-users] Odd one way RTP on SIP to SIP calls

[Julio Arruda]

Just a guess in fact..but..
I'm sure others would love to know how is the NGSS (SST now ?) config 
for this purpose, as well as your sip.conf and etc (one note, you are 
running SN09 or ISN09 ?
Not sure, but this also would help others out there.. :-)



Örn Arnarson wrote:
> Julio,
> 
> It seems you had something going there; I disallowed ISUP messages on
> the SIP-T server and now I have two way audio.
> 
> Thanks a lot for your help!
> 
> Best regards,
> Örn
> 
> On 10/1/07, Örn Arnarson <[EMAIL PROTECTED]> wrote:
>> You are right, the remote server is a SIP-T.
>>
>> I haven't had any problems connecting it to regular SIP servers
>> thusfar though. Also like I mentioned, I don't have this one-way RTP
>> problem with an earlier version of Asterisk.
>>
>> Thanks for your reply,
>> Örn
>>
>> On 10/1/07, Julio Arruda <[EMAIL PROTECTED]> wrote:
>>> Is this a SIP connection or a SIP-T one? Not sure (don't have access to
>>> my previous life docs :-), but this seems to be a Session Server Trunks
>>> doing SIP-T, not sure is the configuration you want...Have you tried to
>>> contact their support ?
>>> PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't
>>> remember seeing in plain SIP calls, so that is why I suspect is
>>> configured as a SIP-T.
>>>
>>> Örn Arnarson wrote:
>>>> Hi everyone,
>>>>
>>>> I'm having an odd problem with one way RTP on SIP to SIP calls.
>>>> I have two SIP servers, one is an Asterisk and the remote SIP server
>>>> is a Nortel SIP server.
>>>>
>>>> When a call comes to the Nortel server through the PSTN and is routed
>>>> to the Asterisk, audio is fine. Two way RTP and no problems. When a
>>>> SIP client registered on the Nortel server calls the Asterisk, the
>>>> Asterisk doesn't seem to send any RTP.
>>>>
>>>> As far as I can tell, there isn't anything wrong with the call setup.
>>>>
>>>> show core version shows:
>>>> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
>>>> 2007-05-17 06:39:34 UTC
>>>>
>>>> SIP and RTP debugging on Asterisk shows this:
>>>> http://www.arnarson.net/~orn/calldebug.txt
>>>>
>>>> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
>>>> root @ build.trixbox.org on a i686 running Linux on 2007-04-25
>>>> 19:59:21 UTC) on the same network (same subnet and physical location)
>>>> as the 1.4.4 this problem does not exist. There is no RTP problem when
>>>> SIP clients registered on Nortel call.
>>>>
>>>> If anyone could help or suggest anything it would be greatly appreciated.
>>>>
>>>> Best regards,
>>>> Örn
>>>> ___


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Re: [asterisk-users] Odd one way RTP on SIP to SIP calls

[Julio Arruda]

Is this a SIP connection or a SIP-T one? Not sure (don't have access to 
my previous life docs :-), but this seems to be a Session Server Trunks 
doing SIP-T, not sure is the configuration you want...Have you tried to 
contact their support ?
PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't 
remember seeing in plain SIP calls, so that is why I suspect is 
configured as a SIP-T.

Örn Arnarson wrote:
> Hi everyone,
> 
> I'm having an odd problem with one way RTP on SIP to SIP calls.
> I have two SIP servers, one is an Asterisk and the remote SIP server
> is a Nortel SIP server.
> 
> When a call comes to the Nortel server through the PSTN and is routed
> to the Asterisk, audio is fine. Two way RTP and no problems. When a
> SIP client registered on the Nortel server calls the Asterisk, the
> Asterisk doesn't seem to send any RTP.
> 
> As far as I can tell, there isn't anything wrong with the call setup.
> 
> show core version shows:
> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
> 2007-05-17 06:39:34 UTC
> 
> SIP and RTP debugging on Asterisk shows this:
> http://www.arnarson.net/~orn/calldebug.txt
> 
> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
> root @ build.trixbox.org on a i686 running Linux on 2007-04-25
> 19:59:21 UTC) on the same network (same subnet and physical location)
> as the 1.4.4 this problem does not exist. There is no RTP problem when
> SIP clients registered on Nortel call.
> 
> If anyone could help or suggest anything it would be greatly appreciated.
> 
> Best regards,
> Örn
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Re: [asterisk-users] Sort of OT: PBX vs CO

[Julio Arruda]

Just one question, why would the Asterisk be involved in the voice path 
at all ?
I would assume a media gateway (TNT ?) would be the obvious choice to 
provide trunking side. And, for line side another gateway (not so sure 
would be as often seen), but in this case a Line side gateway, and 
again, * would not need to be in the voice path.
I know that TNTs with DS3 cards have been used by persons in this 
mailing list, and I assume that some vendor of Line gateways 
(mediatrix/someothers  for low density, I guess some 
genband/calix/occam/whatever for higher density) would not have problems 
getting their gear to work with Asterisk.
The missing piece would be the SS7, that I understand others have used 
with * also here...so...
(regarding VOIP not being an option, I have to wonder, for how long :-)..

NGN vendors don't use a TDM switching fabric (example, Nortel CS2Kc has 
no 'ENET' or anything like that), why would Asterisk need one ..


Alex Balashov wrote:
> On Fri, 10 Aug 2007, Jay R. Ashworth wrote:
> 
>> Short version: There's some hope Asterisk could handle the programming,
>> but the switching fabric simply is *not* up to the task yet.
> 
>And I am not sure that kind of DSP density or CPU-bound framing and
> transcoding is even possible.  At the very least, Asterisk would have
> to have a vast array of rather expensive ASIC cards developed around it
> that would offload a great deal of this functionality;  the dedicated
> DSP support is a good start, but nowhere near where it needs to be.
> 
>> And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
> 
>Yep.
> 
>Asterisk is _NOT_ a switch.  Asterisk is not a transit element. 
> Asterisk is an *endpoint*.  It makes for a nice PBX, feature server, etc.
> Kind of like the BroadSoft, but on a much smaller scale.
> 
>> I hadn't been tracking oSS7 lately; didn't realize that.
> 
>As far as I can tell, it's still pretty useless.  There are a variety of 
> commercial/proprietary SS7 solutions available, though, but I haven't
> tinkered.  And in any case they don't strike me as being able to interface
> with Asterisk.
> 
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: +1-678-954-0670
> Direct : +1-678-954-0671
> 
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Re: [asterisk-users] inband DTMF for g729

[Julio Arruda]

Gang Chen wrote:
> - Original Message - 
> From: "Kristian Kielhofner" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, June 22, 2007 4:32 PM
> Subject: Re: [asterisk-users] inband DTMF for g729
> 
> 
>> On 6/22/07, Gary Chen <[EMAIL PROTECTED]> wrote:
>>> We are using Level 3. At this point, changing carrier is not an option.
>>>
>> Gary,
>>
>>  I use Level(3) with G729a and RFC2833.  No problems, no requirement
>> for inband G729.
>>
>>
>> -- 
>> Kristian Kielhofner
>>
> 
> I can connect to Asterisk IVR using a SIP phone and send RFC2833 with g729. 
> It works fine. But when test call from PSTN to Asterisk, if I set dtmf=auto 
> with g729, I got warning saying something like  * does not support inband 
> for g729 and sutomaticlly switch to rfc2833.  If I set dtmf=g729, there is 
> no warning but I have the same problem. This tells me that Level3 does use 
> inband for g729 or maybe I am doing something wrong .


Have you captured the SDPs ? Or even the RTP traffic (if they are doing 
inband, it would show in the RTP stream). And have you tried 
dtmf=rfc2833 ? (not sure what dtmf=g729 does quite frankly).


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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Julio Arruda]

Richard Klingler wrote:

Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;

I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...



I've a 7912G running with 1.4.x and chan_skinny, and seems to be working 
 just fine (better than 1.2 anyway, the 7912G is not the 'heavy usage' 
phone at home, but still..)


I tried twice to acquire the proper license to upgrade the 7912G to SIP, 
but the order got 'dropped' by the reseller after 2 weeks of 'shipping' 
:-), since 1.4 seems to be handling it just fine, I've moved this to the 
lower priority TODO list.




Hermann Wecke schrieb:

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them 
- check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] They ignore my DTMF!

[Julio Arruda]

Benjamin Jacob wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the 
ulaw codec.




Out of curiosity, there is any 'document' about how RFC2833 would be 
better or worse than SIP INFO ?





Pierre Marceau wrote:


Okay, in the SPA-941 admin I changed:

;DTMF Tx Method: Auto
DTMF Tx Method: Inband

and now it works.

Thanks!
Pierre

 


[EMAIL PROTECTED] 2/21/2007 12:09 AM >>>
  

Pierre,
Thats exactly what  Joanna  said in her reply.
Check the client DTMF settings on your phones.
set it to rfc2833 or out-of-band, whatever the config says.

Grandstream by default have inband DTMF set, and usualy ulaw is 
supported as well, and thats the reason ur grandstream works but 
others dont.


cheerz
- Ben.

Pierre Marceau wrote:

 


Hi Joanna,

Thanks for your reply.

In my mind I think it must be some setting in the client (phone) 
becasue the Grandstream GXP 2000 does work and it is using the same 
sip.conf


Extensions:
6000 is xlite softfone
6003 is Linksys SPA941
6004 is Grandstream GXP 2000
6005 is Linksys PAP2NA

Please have a look at my sip conf and suggest any changes I could try...

[general]
context=internal
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
type=friend
secret=XXX
nat=no
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
canreinvite=no
register=8885551234:[EMAIL PROTECTED]
[atlasvoice]
type=friend
host=proxy.atlasvoice.com
username=8885551234
secret=XXX
fromuser=8885551234
fromdomain=proxy.atlasvoice.com
canreinvite=no
insecure=very
nat=yes
context=incoming

[6000]
[EMAIL PROTECTED]
[6001]
[6003]
[6004]
[6005]
[6006]
[6007]
[6008]


Thanks,
Pierre




  

[EMAIL PROTECTED] 2/20/2007 10:47 PM >>>
 


Hi Pierre,

Just a thought..check your dtmfmode in your SIP client configuration, if
your using inband but your codec is not ulaw or alaw the DTMF tones 
will be
misrepresented and thus will not be recognised due to the audio 
compression,
on the other hand if your phones are rfc2833 and asterisk is set to 
inband

you wont hear anything.

Hope that helps.

Best Regards,
Joanna

On 2/21/07, Pierre Marceau <[EMAIL PROTECTED]> wrote:


  

Hello,

I can call out to the PSTN and talk to people but when I have to 
enter a
dtmf tone in an ivr or voicemail system those systems do not 
recognise that
I have sent a tone. This is the case when I make the call with the 
Xlite
softfone or a regular telephone plugged into a PAP2NA or a Linksys 
SPA941.


However... a Grandstream GXP2000 works just great ???

All are extensions on my Asterisk 1.4 box. I am using a voip trunk 
through

Atlasvoice. All extensions are setup identical in sip.conf.

One last thing, if a system wants me to respond 1 for sales 2 for 
service
I can hit the 1 button quickly 4 or 5 times and the remote system 
will get
it. That does not work for a three digit extension as you may well 
imagine.


Any help would be appreciated.


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Re: [asterisk-users] NAT solutions

[Julio Arruda]

Gordon Henderson wrote:

On Thu, 25 Jan 2007, Yuan LIU wrote:

Thanks for this information.  Does this mean two IAX boxes can talk 
behind their respective NAT's (without any server sitting in voice 
path)?  I'm imagining this:


Asterisk1 <--> NAT1 --- { Internet } --- NAT2 <--> Asterisk2


Using IAX, yes. It's quite straightforward to do. You do need to open 
the IAX port on each NAT device though - this may be called 
port-forwarding, depending on the hardware or its configuration 
interface. Essentially, you port-forward port 4569 from the outside to 
the IP address of the asterisk box on the inside on both sides.


Then have a look at:

http://astrecipes.net/index.php?n=204

To get you going.

Is this the concept of STUN?  Does this also create latency (by adding 
an additional leg in the route), packet loss, even jitter?


STUN doesn't intercept the data. It gives the client device hints as to 
how best to traverse the local NAT firewall.


IAX uses a single port for both commands and data. SIP uses more than 
one and thats when it gets hard as it's easy for a NAT router to track a 
single data stream, but tracking multiple is hard. I have noticed newer 
routers offering SIP NAT traversal though (and the later linux kernels 
claim to be able to do it) I guess, like handling FTP (which also uses 
multiple ports) they are inspecting the SIP packet contents to try to 
work out the RTP ports it's going to use and do the right thing.


I did have issues with a Juniper router recently though - the owner 
claimed it has SIP traversal but it didn't work, but when we turned it 
off and used old fashioned port forwarding it "just worked" ...


My experience with SIP ALG implemented in several routers/modems/NAT 
box/fillintheblanksis not exactly good :-)
I saw many cases where the messing around done by the middlebox break 
either authentication+integrity or even the voice path.
I've not tried the SIP ALG in the iptables modules, but, not sure how 
much better would be :-)..




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Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone

[Julio Arruda]

Doug Crompton wrote:

I am not sure what you are asking? The problem is that rfc2833 does not
play well with the spa-3000 and Asterisk. I am not sure if it is limited
to just the spa3k. There is a bug causing this that has been documented.
Google "spa3000 dtmf bug asterisk" for more info. The bottom line is that
you need to use sip info (inband dtmf) if you desire dtmf transfer to the
other party after the call has completed. Such as you call a bank, or you
call your Asterisk voicemail, or your door lock which is actuated by dtmf.
If none of these are of interest and you would rather have the dtmf
features of Asterisk, then use rfc2833. you can't have both!



SIP INFO is not the same as Inband DTMF, that is why I'm asking.
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Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone

[Julio Arruda]

Doug,
You are saying that RFC2833 somehow doesn't work if you have the 
Asterisk AND at a distinct time (still within the same call), the callee 
to see the DTMF, correct ? Would this be in any case ? (meaning, if the 
voice path is going via the Asterisk or UA to UA directly ?)


I've my spa3k right now somewhat far :-), and I can't test it, but you 
know by any chance if SIP INFO would suffer from the same curse :-) ?
From my limited understand, a big difference in this case is that 
RFC2833 really is "in the RTP stream, but is not voice payload", while 
with SIP INFO, is done 100% out-of-band.




Doug Crompton wrote:

I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa)
I have used newer firmwares but find that 3.1.3 had less echo problems.

Connect a real analog phone to spa3000 fxs.  Call it from another source,
when connected send DTMF tones from that source. You should hear at least
100ms or more of the tone. inband should work. I suspect you are using
alaw or ulaw codecs. There is really no reason to use anything else. When
it does not work you will hear nothing more then a click or an ocassional
to short tone.

Another thing to check is that you should not be using any transfer
options in your dial statement (t or T or other special features.

You really have to listen to this to check it and make changes. Be sure to
restart both spa3000 and asterisk when you make changes. Otherwise you can
get fooled.

If you are making the call from the spa3000 fxo to fxs, you need to have
inband in BOTH.

This is a known bug in Asterisk<>spa3000 for dtmf. I think the problem is
somewhat shared but improvements in 1.4 may gelp or fic the problem. I am
using 1.2 so I cannot answer that.

Basically when using the spa3000 you have to make the choice of wether you
want to be able to use dtmf features (transfer etc.)OR have the capability
to send DTMF to or from the caller or callee. you really can't have both.
Thus inband vs. rfc2833. I chose inband so I can interact with called
ivr's and call in from pstn and access my VM.

Doug


On Fri, 12 Jan 2007, Louis-David Mitterrand wrote:


On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote:

The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things as ivr's.

Thanks for your suggestion. We tried that without success (using firmware
3.1.7(GWc))

Do you think an upgrade to 3.1.10 might be warranted?

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Re: [asterisk-users] SPA 3000 won't relay DTMF to doorphone

[Julio Arruda]

Eric "ManxPower" Wieling wrote:

Doug Crompton wrote:

The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for such things as ivr's.


This will only work if you are using ULAW or ALAW codecs.  All other 
codecs will distort inband DTMF.


I've been using INFO in a SPA 3000 for some time, seems to work fine (is 
not heavy usage, but..)

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Re: [asterisk-users] SIP/TCP?

[Julio Arruda]

SIP over TCP != RTP over TCP
The whole latency deal is much more of a concern in RTP (as well as 
trying to deliver a late packet, that will be not very useful also).

As I understand, MS does SIP/TCP on their LCS or something like that.
Still, not RTP over TCP, since it does not make sense for the voice-path.

Tim Panton wrote:


On 5 Jan 2007, at 23:22, Yuan LIU wrote:


From: "James R. Stevens" <[EMAIL PROTECTED]>

TCP is a connection oriented protocol ..as others mentioned, it 
superiority comes because it knows when packets are dropped to resend 
them. It also has mechanisms for flow control etc.. SIP is a 
connection-less protocol. It uses 'best effort' transmissions..if u 
want its delivery guaranteed you must encapsulate it.


So I take it that UDP is just a decision due to popular demand; timing 
(jitter) is a frequently cited factor to favour UDP.  Is there any 
technical difficulties in implementing SIP/TCP within Asterisk?


The reason I'm asking is that there are products that support both UDP 
and TCP.  And SIP/TCP, RTP/TCP have their own merits.


Granted, SIP is connectionless.  So is HTTP (well, for its original 
design anyway).  I notice that guaranteed delivery could be a good 
thing for SIP in many situations; there have also been advancements 
that make timing less an issue in RTP/TCP.


Is "switching to" SIP/TCP - RTP/TCP as simple as rewrap 
messages/streams, or is it more involved?



It's a latency thing.
Say you send a packet every 20ms
Say you have a pair of endpoints with 100ms between them.
Say you drop a packet in the media channel.

TCP will re-request the missing packet, which has 2
bad effects:
1) when it does turn up the packet will be >100ms late so you have 
to play silence

or make something up until it does.
2) all the subsequent packets will be behind the re-tried packet and 
also 100ms
late. - Note that because TCP is a stream, you can't get at the 
subsequent packets

even if they turn up on time because you have to wait for the missing one.
Even more frustratingly you now have to dump these 4 perfectly good 
packets.

If you don't you will have introduced 100ms of lag in the audio stream.

- Of course none of this applies if you are on a LAN - with <1ms 
roundtrip time

as the retry can get to you soon enough to be useful.

With UDP you simply make something up for the missing packet and carry 
on when

you get the next one. - so you make up a single packet instead of 5.

With TCP the lost packet is multiplied by the ratio of the roundtrip 
time to the

packet interval.

Of course you can cover this up by increasing the buffering, but then
you are introducing yet more lag.

So, I simply don't think that TCP is suitable for telephony media 
streams over any
network where the roundtrip time is of the same order as the packet 
interval.


Now there are 'reliable datagram protocols' ( IL for example) but they 
aren't

much used on the internet.


Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Julio Arruda]

Darrick Hartman wrote:

Kenneth Padgett wrote:
I'm looking for opinions on the "best value" router to use for home 
offices.

 It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the phones always have higher priority 
traffic
than the PCs. (and not rely on the phones to do the QoS because some 
PCs may

not be connected to the phones).


I'm using a Linksys WRTSL54GS and 3rd party firmware with great
results! You won't find QoS features in the default Linksys firmware
though, so if you want something out of the box, this isn't much help.

My main reasons for picking it over the older WRT54G's where:

1) It was (still is?) available in retail stores, whereas the WRT54G's
that run Linux are generally only on ebay these days, they have to be
older versions.


Linksys listened to demand and released a version that still has the 
Linux firmware.  The model is WRT54GL with the L for Linux.


Newegg has them for $57 after MIR.


Note, seems like the 'current' WRT54GL has less flash/memory than older 
releases of WRT54GS, sadly :-)
http://wiki.openwrt.org/TableOfHardware?action=show&redirect=toh  may 
help on this case.
Another point is, he mentioned some WRTSL54GS, that seems to be a Linux 
version of the WRT54GS (and seems to keep the same memory/flash as the 
old WRT54GS). Bottom-line, homework is required :-)
I've migrated a pair of these to openwrt (from Talisman) recently, and 
still need to move the home Internet link into one, but it seems quite 
flexible (even some asterisk and openser/related packages are available).
Regarding QoS, as many mentioned, most of your problems will be in the 
upstream (and as a matter of fact, in a residential broadband, is how 
far you can go regarding 'control').
From my experience, one big deal is shaping, you need to configure your 
'QoS border' box to shape the upstream traffic, then I would be 
concerned with priority being given to the VOIP traffic.
Doesn't seems like many used a plain approach of using DSCP for QoS in 
these kind of setups, but many VOIP devices seem to use these just fine 
to tag the packets by default, so I guess this would be a nice way of 
being 'vendor/configuration agnostic', not depending in UDP port ranges 
and etc for classification. When I finish the migration (need to 
schedule some maintenance windows still with wife and mother-in-law), I 
will start to play with distinct devices and etc.
PoE of course is not an option seems in this range of devices 
(price-wise), but..guess is a matter of time :-)


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Re: [asterisk-users] No caller ID, no incoming call

[Julio Arruda]

Try to search for the PrivacyManager application.
It does 'check' if the CallerID is present, if not, it will play an 
announcement to ask the person to 'type' their phone number, and it will 
 allow you to then accept it.



je . wrote:

Is it possible to reject all incoming calls that do
not have a CID?

Could I do something like that (modified version from
the book): 


exten => 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10)
exten => 123,10,Dial(Zap/4)
exten => 123,20,Playback(abandon-all-hope)
exten => 123,21,Hangup(

Alternatively, what's the privacy.conf file for? What
does it mean for a user to have to chances to 'enter
his CID' else his call is rejected?


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Re: [asterisk-users] Re: G722?

[Julio Arruda]

Benny Amorsen wrote:

"MG" == Michael Graves <[EMAIL PROTECTED]> writes:


MG> Who will benefit as long as calls must typically pass into
MG> existing PSTN infrstructure, and so be transcoded into G.711? It
MG> seems to me that only systems that are IP end-to-end stand to show
MG> the improvements...or am I mistunderstanding?

ISDN can transport G.722. Can Digium PRI cards do G.722?


Still, I think his point is the weakest link still the PSTN hop, no 
matter where it will happen.
If you had only VOIP end-to-end, G.722 would be good, but in any step 
going via PSTN (the PRI idea only solve one side of the PSTN 
call..unless you have the PRI in both ends of the call with G.722 ?)
As people move forward into VOIP and peering, G.722 (and others) will 
come into play I guess, meanwhile, only for interoffice VOIP calls I guess ?

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Re: [asterisk-users] Re: VOIP Bandwidth questions

[Julio Arruda]

Eric "ManxPower" Wieling wrote:

Vikki wrote:

I think vonage is using g723.1 which requires 6.4kbps voice bandwidth
compared to g711 - 64kbps.

For SIP to SIP calls, RTP doesn't necessarily goes thru the server. Only
Signalling goes to the servers. This means no bandwidht usage for the
provider.
For SIP to PSTN calls, it has to goes thru a media gateway (owned by the
provider) which may be seperate from the sip server.


I imagine that most of Vonage's customers are behind NAT and direct RTP 
(re-invites) don't work well with the endpoints behind NAT.


At least some solutions for 'remote end NAT'(not 'CPE solved the 
problem' NAT) can involve 'distributed' RTP Proxies, to anchor the RTP 
call legs.


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Re: [asterisk-users] Re: Re: FW: Peter Dicks Chairman ofSportingbet PLCisarrested at JFK!!

[Julio Arruda]

Steven wrote:

Because the Telco is government owned.
They are the PSTN, so only they can route and charge for PSTN calls.

Making a call from an Indian office to a US office over VOIP is legal.
Forwarding a PSTN call over that same VOIP trunk is illegal.



In other countries where the Telco is not government owned, still 
similar constraints exist, where some special license is required..
The folks in Brazil should know better than me, but also I remember some 
other Latin America countries in the same kind of situation.


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Re: [asterisk-users] SNOM 360

[Julio Arruda]

Dovid Bender wrote:


- Original Message - From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360



On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:

On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:

> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets real bad. The docs from SNOM
> on TOS (or DIFFSERV) is poor and I dont understand it well enough.
> Anyone have configs or docs on how they did this ?

I would be surprised to learn that the Snom is actively doing traffic 
management itself.
Traffic managment must be done at the bottleneck to be halfway 
successful. Let's
assume you are doing a download and you snom would do traffic 
management giving
itself priority. What if your co-worker is doing a huge download? How 
should your

snom know and throttle his download? No way.


That is a different problem entirely, and as you say, the snom cannot
do anything about a remote bottleneck (except perhaps theough QoS and
TOS flags in the data it sends).

The snom does seem to manage its two local ports properly though but
this cannot be hard. Worst case is that the snom needs about 128Kb/s -
Not hard on a 100Mb/s full duplex connection :)

Dovid - Have you identified where the bottleneck is in this case? You
do not specify as far as I can see. Is the VoIP call using the
internet, or is it local?

Regards,
Steve
It is using the internet. The problem is when a user starts a big 
download. The phone call goes to s***.



Dovid,
I would guess that:
First thing would be quick&dirty ASCII drawing, showing where is the PC, 
the SNOM and the "sources/destinations" of the Internet and VOIP traffic.
You mentioned download, assuming this is a DSL connection, this would 
be, when it arrive at the IP phone, would be too late to do anything, IF 
you are bumping into a bottleneck in the DSL downstream.
What 'direction' of the voice path is suffering, did you capture the 
traffic (is it suffering because of jitter, packet loss, ...) ?
Like others mention, QoS (the buzzword :-), is a very wide and generic 
term, and you will need to 'isolate' the problem to see if a solution is 
feasible.


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Re: [asterisk-users] Re: Re: Re: TE420P/TE415P?

[Julio Arruda]

Matt Florell wrote:

Yes, that is very confusing :)

Is there no way to throw a timer chip in there(I suppose it's way too
late to put that suggestion forward now)?


Curiosity, isn't the timer from the 2.6 kernel 'good enough' for 
Asterisk purposes nowadays ?
Or there is a constraint using 2.6+ztdummy that is not obvious (to me at 
least :-)) ?




There are many instalations that need a zaptel timer to use meetme
with no telco lines and this will be the first Digium zaptel card made
that does not provide such a timer.

Even if it was just a fake X100P-type single FXO channel that is not
used, it would still be a wonderful addition(and relatively cheap) to
fully utilize Asterisk with only VOIP channels. Otherwise there are a
lot of installations that will still have to have that extra X100P
card in their machines.

Thanks,

MATT---


On 7/31/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:

- Matt Florell <[EMAIL PROTECTED]> wrote:
> Will this card serve as a zaptel timer?
>
> It's kind of hard to imagine that it wouldn't, I just want to make
> sure since this is the first Digium card with no telco interfaces.

No, the transcoder card is not a Zaptel interface, so it does not 
provide any 'spans', and thus cannot provide timing. The interface to 
it is part of Zaptel, though, just to keep everyone confused :-)




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Re: [asterisk-users] SRTP enabling

[Julio Arruda]

Abdul Lateef wrote:

Hi everyone,

I was trying to support SRTP in asterisk for our
Linksys IP Phones to prevent of ISP blocking issue.

I compiled successfully SRTP from
http://srtp.sourceforge.net/srtp.html 
But i don't know from where i should start to

configure in Asterisk.

Could someone please give me the example sip.conf for
the way how i can support?

You replies will be high appriciated.


Most of the blocking in other countries, was not for RTP traffic, but 
for signaling traffic (SIP usually, Mexico x Vonage comes to mind).

You are sure they are blocking RTP traffic ?
And, from what I understand, in some places the gov. forced the ISPs to 
remove the blocking (at least, I heard of one such a case in Brazil, a 
DSL provider started to block SIP, and Anatel, Brazil gov. entity that 
regulate telephony and others, asked them to remove the blocking, others 
with more knowledge of the case may be able to add their remarks)


Blocking SIP if you control the server is somewhat easy to prevent (if 
is a plain dumb UDP port 5060 filtering), just have your server listen 
in another UDP port...


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Re: [asterisk-users] Tadiran Coral IP PBX to Asterisk

[Julio Arruda]

Bill Gibbs wrote:
Goal – to get the CoralIP PBX long distance savings by sending it to 
Asterisk (which then talks via SIP to other long distance voip providers)
The Coral IP supports MGCP and so does Asterisk.  Has anyone tried 
sending calls from the Coral PBX to Asterisk via MGCP?  I will be 
playing around with that this weekend but thought I’d ask.
The other way I was thinking was doing a back to back PRI, utilizing a 
Digium TE110P.  If I understand that process correctly, using back to 
back PRI cards (one in the Tadiran and one in the Asterisk server) we 
can basically open a digital trunk to send (and accept) the calls.
Any suggestions on integrating a non SIP (but VOIP) style PBX to 
Asterisk other than what I outlined above?


From what I understand, you can connect MGCP clients to Asterisk, so, 
if your Coral IP PBX is "a mgcp client", it may work, otherwise...
(quite frankly, I don't think it is the case, from wa quick search in 
google, it seems the Coral IP PBX is a Call Agent itself, not a media 
gateway/MGCP client)


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Re: [Asterisk-Users] Motorola and Asterisk

[Julio Arruda]

Most likely, he is thinking something like using the MTA (a motorola 
cable modem with RJ11 phone ports), to register to Asterisk.
From what I understand, most (if not all) packet cable VOIP is done 
using NCS (a mgcp-like protocol ?) as call control, not SIP.



Alexander Lopez wrote:

Isn’t DOCSIS a network layer 1 or 2?

I

TCP/ip would run on top of a DOCSIS network

SIP on top of TCP/ip

 

 

DOCSIS specifies downstream traffic transfer rates between 27 and 36 
Mbps over a radio frequency (RF) path in the 50 MHz to 750+ MHz range, 
and upstream traffic tranfer rates between 320 Kbps and 10 Mbps over a 
RF path between 5 and 42 MHz. But, because data over cable travels on a 
shared loop, individuals will see tranfer rates drop as more users gain 
access


 

 


So take your DOCSIS standard device and plug it into your * box.

 

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Carlos 
Alberto Bernat Orozco

*Sent:* Sunday, July 02, 2006 9:51 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Motorola and Asterisk

 


Hi Group

Does anybody knows if Asterisk have plans to work with Motorola and 
DOCSIS? I'm trying to make work SIP into an PacketCable arquitechture 
but I can't figure out with Asterisk.




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[Asterisk-Users] SPA 3102 Caller ID in Bellsouth/NA

[Julio Arruda]

Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ?

From a quick test (got mine yesterday), seems like it is not 
recognizing Caller ID from PSTN/FXO port..
Using the  same configuration as a Sipura 3000 (to be sent to 
mother-in-law POP :-), no Caller ID at all, (I've even extended the PSTN 
delay to give it more time, but no dice).


www.voxilla.com forum has a couple of posts with the same results also, 
so I not sure is my mistake as usual (or at least as my wife want to say 
:-)..

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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

[Julio Arruda]

Peter Bowyer wrote:

On 22/05/06, Steve Kennedy <[EMAIL PROTECTED]> wrote:


On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:

> If you want to roam between GSM and WiFi while on a call, the GSM
> carrier is going to have to support it.

There is a protocol for this (UMA), however few operators as yet support
it.

T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed
monthly fee), but they are are going to (if not already) block VoIP
calls - they've realised that users are using VoIP (probably Skype) and
not making GSM voice calls - and the voice revenue is declining.



They block VoIP and IM, supposedly to protect their users from a poor
quality experience. Of course, it's really to protect their voice and
SMS revenues.


From what I understand, T-Mobile UK just announced they would block 
VOIP earlier this month, but that is quite recent, and I don't recall 
seeing this 'globally' announced.



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Re: [Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL

[Julio Arruda]

There is a mailing list (in portuguese, but most persons there will 
answer your questions in english without problems) in Brazil.
IF your question is biz related, of course, there is a proper place for 
these (a biz list).
http://listas.asteriskbrasil.org/mailman/listinfo would give you both 
lists (the tech.list and the biz list).



Tele Cost Price Reducer wrote:

hi,
you can try the following:

exten => s,1,Gotoif($[${CALLERIDNUM} = 1130851536 ]?10)
exten => s,2,Goto(from-pstn,s,1)
exten => s,10,disa(no-password,from-internal)
* *
it works for me.
if you need further help, let me know.
BTW, i am very interested in the Brazilian Market so i would like to get as
much as possible info about Brazil.

good luck,

Mickey

On 5/6/06, ITN Info - 11-3898-0112 <[EMAIL PROTECTED]> wrote:



 Hi,



I am trying to create a situation where I call the DID number which is
1140636249 and I receive a dial tone to dial. I d like also to 
autenticate

the number 1130851536.

I can see that asterisk decode this number but I dont know how to
authenticate this number only. This is what I am doing



*Sip.conf*



[globo]



type=friend

username=itn111

fromuser=itn111

secret=123456

insecure=very

host= globo.net.br

context=fromttt

fromdomain= globo.net.br

dtmfmode=rfc2833

disallow=all

allow=g729



register => itn111:[EMAIL PROTECTED]:5060/itn111



where itn111 is the LOGIN for DID and the virtual extension for *
Extensions.conf *file



*Extensions.conf *



[fromttt]



exten => itn111,1,Dial(SIP/29650,60,Ttr)

exten => itn111,2,Hangup()



This settings above can can garantee that every call to 1140636249 
goes to

extension 29650. Do DID part is working ok.

Now I would like to get a second dial tone when I call 1140636249 for
asterisk DISA.

I d like also to autenticate the number 1130851536 (caller number) and
only this number can receive the call



This is what I am trying to do



exten => itn111,1,Dial(SIP/29650,60,Ttr)

exten => 29650,2,DISA(no-password|brasil) ; I use no-password for this 
for

now and Brasil context

exten => 29650,3,Hangup()



or



exten => itn111,1,DISA(no-password|brasil)



In sip show channels I see



*SIP/itn123456-cdfe� (fromgvt��� itn123456��� 1�� )� Up DISA�
no-password|brasil*

* *

But there s no dial tone. And I don t know how to authenticate this 
number

1130851536. I see that asterisk collect this number



Can you pls help me to do this settings ?







Atenciosamente







*Diretoria Comercial - Newton Medina*

*PABX11.3085.1536*

*MSN [EMAIL PROTECTED]



Rua Augusta 2.212 SL 26 Jardins 01412001

S�o Paulo - Brasil





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Re: [Asterisk-Users] Setting Codecs on the Fly

[Julio Arruda]

Douglas Garstang wrote:

Does anyone know if it's possible to set the codecs for a number via an 
Asterisk command?

Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a 
command that can set the same thing so that it can be done without having to 
change sip.conf.

Essentially I want the user to be able to prefix a code to their dialled number 
to set their preferred codec for that call.

Possible?


Humm..I wonder if what google returned for:

"asterisk set codec on a call"

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

Would help...Seeems that in fact, google is my friend:

"${SIP_CODEC}: Used to set the SIP codec for a call"
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Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?

[Julio Arruda]

Paulo Scardine wrote:
I have a worst issue for you... If your fax solution is ever going to 
receive fax in Brazil, how would you block collect calls?
I have made a fax server solution with cheap Digium hardware that works 
in Brazil (2 E1s).

--


Paulo,
He is mentioning E1/PRI, so I assume the well known "collect call on 
E1/R2 thingie" doesn't apply to him.




Adolfo R. Brandes escreveu:


Greetings, All-Knowing Asterisk Users List,

My company needs to build a reliable fax server that can handle at 
least 30 simultaneous incoming faxes from the PSTN, using PRI.  We 
realize that this can be solved in any number of ways using a Linux 
box, but since IVR is also a must, Asterisk popped up as the most 
promising solution.


After combing these lists for clues, we began experimenting 
extensively with Asterisk and its software DSP and fax capabilities in 
most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, 
together with Digium's E1 cards in server-grade Intel motherboards, 
all in a dedicated test environment.


Unfortunately, though, we have yet to achieve reliable and 
satisfactory results, even with only 1 fax call at a time.  I won't go 
into the details because we don't need technical support, given that 
this is, as of yet, a very loosely defined test.  What we want is is 
merely a pointer in the right direction. So here it comes:


Has anybody ever achieved, or know of someone who has, reliable 30 
simultaneous PRI fax calls using Asterisk and Asterisk-compatible 
hardware and software?


We are hardware agnostic, so if you say Sangoma's cards do it 
better than Digium's, or that Eicon Diva cards' hardware DSP and 
chan_capi are the only solution, we have no problem going there.  I 
would be most thankful, however, for detailed explanations of 
successful scenarios, including such things as motherboard make and 
model, processor speed, Linux distribution and version, and anything 
else you decide to be even marginally pertinent.

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Re: [Asterisk-Users] SIP through freeBSD NAT

[Julio Arruda]

Eric "ManxPower" Wieling wrote:

Use a codec your phone supports like ulaw.



Assuming he is using SJphone, that I understand, would support iLBC even 
in the free version ?




Alyed Tzompa wrote:


made the changes in sip.conf so now it reads:

disallow=all
allow ilbc

now I when the call is placed it is not hanged up, but I cannot hear 
anything. I think it's becasue Asterisk is sending the RTP's to a 
wrong address (my

internal IP).
Looked at the sip debug and got the following:

-- Executing BackGround("SIP/alyed-5a8d", 
"/var/lib/asterisk/sounds/testt") in new stack

We're at 200.78.243.12 port 13458
Answering with preferred capability 0x400(ILBC)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
90.0.0.10;branch=z9hG4bK5a0a00c043bab4f9390f1bef02ef;received=201.127.53.246;rport=5060 


From: "unknown";tag=2438130825771721203
To: ;tag=as7222f729
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 17028 17028 IN IP4 200.78.243.12
s=session
c=IN IP4 200.78.243.12
t=0 0
m=audio 13458 RTP/AVP 97 101
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 201.127.53.246:5060
-- Playing '/var/lib/asterisk/sounds/test' (language 'en')
Integra2*CLI>

Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 
90.0.0.10;rport;branch=z9hG4bK5a0a00c043bab4f944b4f6f302f2

Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
From: "unknown";tag=2438130825771721203
Max-Forwards: 70
To: ;tag=as7222f729
User-Agent: SJphone/1.60.299a/L (SJ Labs)


9 headers, 0 lines



any ideas?




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Asterisk Users Mailing List - Non-Commercial Discussion 


Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
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Alyed Tzompa wrote:
 > sip.conf
 > [general]
 > port=5060
 > externip = www.theip.net
 > localnet = 192.168.1.0
 > localmask = 255.255.255.0
 > allow=all

Don't use allow=all. Use disallow=all and then allow= line for the
specific codec you want to use.

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Re: [Asterisk-Users] Translating between different codes

[Julio Arruda]

From what I can see
The 2 legs of the call are: 'phone' in alaw and 'laptop' in g726, why 
should he need G.729 anywhere ?


Bartosz, not exactly that familiar, but I guess you could try to debug 
the call establishmment.
(one thing that puzzles me, you mention "IAXy", but you show 2 sip.conf 
entries..should not be one in iax.conf and one in sip.conf ?

(of course, with the proper syntax for each .conf file).

Moises Silva wrote:

be sure you allow the g729 codec in [general] context in sip.conf for
the sjphone.

On 1/2/06, Bartosz Wegrzyn - asterisk <[EMAIL PROTECTED]> wrote:


Hi,

I would like to know if asterisk is able to translate between two
differnet codecs. For example:

I have this config in sip.conf file:

[phone]
disallow=all
allow=ulaw
dtmfmode=rfc2833
dtmf=rfc2833
username=phone
type=friend
host=dynamic
secret=
mailbox=3001
context = sip
callerid="Wireless <3001>"
canreinvite=no
qualify=yes
qualify=3000
nat=yes

[laptop]
disallow=all
allow=g726
dtmfmode=rfc2833
dtmf=rfc2833
username=laptop
type=friend
host=dynamic
secret=
mailbox=3002
context = sip
callerid="Laptop" <3002>
canreinvite=no
qualify=yes
qualify=3000
nat=yes

Should asterisk translate between two codes.
First clent is iaxy, second is sjphone.
It is not working for me, and I am getting error on sjphone:
"Unabke to agree on media streems".

When I change the codec for laptop to ulaw everything worls ok.
This would mean that asterisk cannot establish communication if both ends
have different codecs supported. Is this right???

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Re: [Asterisk-Users] Semi-OT: porting numbers away

[Julio Arruda]

Since the last hurricane (that left me without phone for around 3 weeks 
or so), I did the call forwarding (remote call forwarding in fact).

Lucky I was running in the cable modem in a couple of days (power restored).
I was planning in having two DIDs in distinct providers (I've been using 
them for outbound in BYOD contracts), and keep the "POTS number" in the 
more stable one.

But I'm still concerned with the 911 issue.
Is the POTS telco (Bellsouth in South Florida in my case) mandated to 
provide 911 in a "ported line" ? I'm not that confident in using 911 via 
the ITSP.



Kerry Garrison wrote:

-- Personal opinion alert --
 
Do not route everything to an ITSP. At minimum keep a main PSTN line with

call forwarding or call forwarding on busy until you are 1% confident
that the service works, is reliable, stable, and will have some staying
power.
 
Kerry Garrison

Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 -  
[EMAIL PROTECTED]
  http://www.techdatapros.com 

  _  


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross C
Sent: Thursday, December 29, 2005 5:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Semi-OT: porting numbers away



I'm looking to move one of my clients to an Asterisk system and a VoIP
provider (Teliax, Voxee, ViaTalk, Voicepulse).  My concern is porting my
client's numbers to a VoIP provider.  Let's say we get all their numbers
ported to Teliax (or Voxee or viatalk, etc.), everything is peachy for a
year, then Teliax gets sued for some reason or another, and goes bankrupt
and closes its doors.  That, obviously, leaves my clients without phone
service.but what happens to their numbers?  If the VoIP provider goes out of
business, can I go to another VoIP provider or a ma bell and transfer the
numbers to them even if Teliax (or whomever) is unreachable and off the map?

 


Thanks in advance for any info!

 

 


-Ross






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Re: [Asterisk-Users] Bonded ethernet ports and *

[Julio Arruda]

Rich Adamson wrote:
..

A fairly common assumption is the failover happens in xxx milliseconds, but
due to nic card design (etc) a different MAC address is used in the failover
condition. That confuses the hell out of the layer-3 boxes and negates the
value of the failover. (All documentation, etc, is correct but actual
implementation in this example is limited by the nic card's inability
to use a different MAC address from what's programmed into it. There are
a large number of current nic cards like that.)



I can see this being a problem (not only with layer 3 boxes, but even 
with other IP endpoints in the same segment, after all, their ARP 
binding will become useless in this case..

But would you have any example of this ?
Most of the time, should be a no-brainer to have the NIC use a 
"non-burned-in-address", in fact, I wonder if is not required by the 
802.3 specs (LAA), since protocols like DecNET, SNA and etc tend to 
require this.


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Re: [Asterisk-Users] Bonded ethernet ports and *

[Julio Arruda]

Rich Adamson wrote:
.


Last, the bonding of two nics at the server level _requires_ the associated
switch interface to support the exact same bonding algorithm. Historically,
that has been a problem for many switch vendors.


Not so sure I understand, but if you mean, 'the algorithm to select a 
link(usually a hashing of some layer source/destination to ensure 
sequence)' ? This would not really to match in both ends, AFAIK.


In some implementations I've worked with, the non-use of proprietary 
protocols to 'establish/maintain' the LAG/whateveriscalledit group would 
force you to use 'static' assignments of the members, but other than 
that, not big deal..(counting on RFI for failure detect and etc helps...)



Short answer... I'd never do it. Long answer... think in terms of high
availability "systems"; the nic card is the least concerning.


There are quite few carriers using similar 'bonding' to dual core 
ethernet routing switches doing "split-MLT", where the 2 chassis would 
'look like' a single box with bonding/FEC/MLT links, so is more than NIC 
card only. I would go as far as say that most of the time is done for 
redundancy, not for bandwidth (call signaling and announcements only 
voice bearer)


Anyway, it would seem to me the original poster was looking for 
redundancy, not really 'added bandwidth'.

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Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline

[Julio Arruda]

I was testing Broadvoice few weeks before Hurricane Wilma here in FL.

Since then, I had been since the landline (Bellsouth), and I had to 
'remote callfwd' the BS # to my broadvoice #.


So, from my impression, is ok for my needs (I got a weird no ringback 
problem that I kind of solved with a "Background" trick), and no 
surprises yet regarding the bill (my mother in law call Brazil a lot 
from my house, no, she is not aware of the 'unlimited' plan. So I may be 
in for a surprise in a couple of months).

I've no tried several calls at the same time, you may want to ask them..
PS: I'm running Asterisk 1.0.9

Dane Reugger wrote:

We are considering Quantumvoice as a provider -

They are telling us they will give us 1 line number but we can have 5 
concurrent incoming and outgoing line numbers. Charge is about $45 + 
extras - this seems considerable less expensive than the competition 
which seem to focus on.


My second choice is BroadVoice $29.99 + $9.99 per additional line (in 
state only?) - more expensive, less features, and they don't seem loved 
by many ?


Is anyone else using Quantum Voice?
It was mentioned earlier that it requires an ATA connection and Asterisk 
support/compatibility is sketchy at best - I've contacted BV and they 
responded saying they need 24hrs to look into it?


Seems like a popular topic but I'm looking for 2-3 lines - I only need 
one number but need to be able to make or receive several calls at a time?


Any advice or recommendations appreciated - I want to port  my number 
but I'm running out of time and must make a decision very soon.



Thanks,
Dane Reugger
Crescent City Technologies
New Orleans, LA 70112
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Re: [Asterisk-Users] R2-Digital (Q.421)

[Julio Arruda]

Just to clarify this in my head :-)..

So...
They are using E1/R2 (the R2 Digital)in fact, for all the line signaling 
 (nothing unusual)
The register signaling, that I was under impression would be MF in each 
timeslot (MFC5C in .br, not sure if the same in others), is in fact DTMF 
in this trunk, and only to provide DNIS ?
(in Brazil R2, the register signaling has some collect call information 
and etc).


Steve Underwood wrote:

Hi,

I tried hunting for a little more info. I think all that happens with 
this is they use the Q.421 spec for handling the ABCD bits, and then 
simply send the DNIS through as DTMF after the seize if acknowledged. 
That means they loose some of the functionality of real R2 signalling - 
e.g. no busy, NU, or congestion detection. It wouldn't take a lot of 
work to implement that.


Regards,
Steve


Steve Underwood wrote:


Hi Jesus,

The Cisco kit, and one or two other products, offer an R2 digital 
using DTMF mode, but this is the first time I have heard of it being 
used. The spec for this is definitely not Q.421. That spec does not 
mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU 
specs, as far as I can tell. Without a spec, or any equipment to play 
with, there isn't a lot I can do right now.


Steve


Jesus Mogollon wrote:


Hi Steve:

 Thanks for your help. I really appreciate it..

  My provider is CANTV in Venezuela. There's a venezuelan variant in 
the code and I'm using that. Incoming works perfectly, outgoing is 
not working. I'm being told that incoming is MFCR2 but outgoing is 
R2-Digital with DNIS DTMF. There is a Cisco router working and it's 
using the following:


r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED] 
>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of
your
channels, and providing all your incoming calls on those channels.
It is
use FX signalling for other channels, and you must make your 
outgoing

calls there. Someone else told be about a similar configuration. I
think
they were able to use chan_zap for the other channels, and make
use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:

> Steve:
>
>   That's exactly what I'm using. Incoming calls work like a
charm but
> when I try calling I get a protocol error. My provider says 
that for

> outgoing I need to use fx signalling. I see that in unicall.conf
> there's such a thing as protocolvariant=fx but if I uncomment that
> line, unicall gives me an error. Any ideas? Thanks for your 
help...

>
> 2005/11/4, Steve Underwood <[EMAIL PROTECTED]

> mailto:[EMAIL PROTECTED]>>>:
>
> Jesus Mogollon wrote:
>
> >Does anyone know how to make this work with Asterisk?
(R2-Digital
> >(Q.421)) I have MFCR2 configured but I'm told that outgoing
calls are
> >to use Q421 R2 Digital signalling. Any help is appreciated.
> >
> >Jesus Mogollon
> >
> >
> See http://www.soft-switch.org 
>
> Steve



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Re: [Asterisk-Users] R2-Digital (Q.421)

[Julio Arruda]

Jesus Mogollon wrote:

Hi Steve:

Thanks for your help. I really appreciate it..

My provider is CANTV in Venezuela. There's a venezuelan variant in the code
and I'm using that. Incoming works perfectly, outgoing is not working. I'm
being told that incoming is MFCR2 but outgoing is R2-Digital with DNIS DTMF.
There is a Cisco router working and it's using the following:

r2-digital-dtmf-dnis R2 ITU Q421 DTMF tone signaling with DNIS


I may be a little lost here, but isn't MFC5C E1/R2 just plain R2-Digital ?
The fact that you have DNIS or not doesn't change it (in brazil was kind 
of weird, you had to have the proper number of digits or something like 
that), but from my memory:


E1/R2 - Line Signalling - TS 16, and ABCD bits
Register Signaling - Inband - with DTMF fwd and back (Compelled I thing 
was the term used).





What's the equivalent in libmfcr2 and Unicall?

Again, thank you for your help and your code!

Jesus Mogollon

2005/11/5, Steve Underwood <[EMAIL PROTECTED]>:


Hi Jesus,

FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of your
channels, and providing all your incoming calls on those channels. It is
use FX signalling for other channels, and you must make your outgoing
calls there. Someone else told be about a similar configuration. I think
they were able to use chan_zap for the other channels, and make use of
its FX signalling features. I am not sure how that works, as FX
signalling over E1s is far from standardised.

Regards,
Steve


Jesus Mogollon wrote:



Steve:

That's exactly what I'm using. Incoming calls work like a charm but
when I try calling I get a protocol error. My provider says that for
outgoing I need to use fx signalling. I see that in unicall.conf
there's such a thing as protocolvariant=fx but if I uncomment that
line, unicall gives me an error. Any ideas? Thanks for your help...

2005/11/4, Steve Underwood <[EMAIL PROTECTED]
>:

Jesus Mogollon wrote:



Does anyone know how to make this work with Asterisk? (R2-Digital
(Q.421)) I have MFCR2 configured but I'm told that outgoing calls are
to use Q421 R2 Digital signalling. Any help is appreciated.

Jesus Mogollon




See http://www.soft-switch.org

Steve



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Re: [Asterisk-Users] DNS Server Failure wreaks havoc

[Julio Arruda]

DNS caching server running in the same machine ?

Eric "ManxPower" Wieling wrote:

Um, put in IP addresses instead of hostnames in Asterisk's config files?

Eric Bishop wrote:


I agree about Asterisk being terrible with DNS failure, but how can you
avoid using DNS on *nix system?

On 11/7/05, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:


Brian Capouch wrote:

I don't think this is a new issue--I've seen it talked about on the 
list

before. I don't know if I've ever seen anyone post a fix.

My DNS server went out last night in a horrendous storm when an 
upstream

link went down. The madness is that the behavior of the whole server,
including the part that's handling my POTS lines, gets wigged out on a
DNS failure, making the whole system unusable. I have two questions;
being able to solve either would be wonderful:



Asterisk is horrible at handleing DNS failures. Don't use DNS with
Asterisk.


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Re: Asterisk + 99.999s was (Re: [Asterisk-Users] Asterisk on windows)

[Julio Arruda]

Patrick wrote:

On Sat, 2005-10-01 at 08:31 -0400, Julio Arruda wrote:
[snip]

One thing interesting, coming from data background, seeing the 
requirements in carrier voice networks. Is a quite distinct ball-game.
Devices that require 'hot-software-upgrades', still not that often seen 
in data. How is this being handled with Asterisk + other solutions ?
Example, having a trunk gateway with a OC3 worth of TDM, is 'acceptable' 
that a sw upgrade will cut established calls ?



Iirc Motorola has a solution that allows in-operation linux kernel
upgrades. No idea how they pulled that magic off (and if it actually
works). At VON IBM was going to demo a blade based Asterisk solution
that has auto-failover of calls so maybe that could also be used to
upgrade software. Don't have more info about this IBM solution. If you
have a DS3 or OC3 worth of TDM calls then it probably makes sense to use
a carrier-class box.


Weird as it seems, not sure if the softswitch itself is the problem.
Example, you could have a media gateway where the established calls are 
not torn down during a software upgrade of the Media Gateway controller 
'entity'.
The hardest part is the media gateway failover, I'm only familiar with 
Nortel (I work in Nortel) MG, and they in some cases would do these with 
APS and 1:1 sparing of the cards, where the sw migration is a 'hitless 
process', I assume others have similar options, but again, is not 
exactly 'in the asterisk' only, is in more than that, is in the 'solution'.





Regards,
Patrick
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Asterisk + 99.999s was (Re: [Asterisk-Users] Asterisk on windows)

[Julio Arruda]

Patrick wrote:

On Wed, 2005-09-28 at 23:17 +0800, Steve Underwood wrote:
[snip]

An effective DOS attack on a $300,000 Alpha running NT I used to use was 
"wiggle the mouse" :-) I never really understood how that brought a 
multi-CPU machine to a standstill, but it did.



Reminds me of an Internet Call Diversion pilot WorldCom did back in 2000
where Alcatel & some M$ drones brought in 2 very big Alpha servers
running NT. These boxes needed to be rebooted multiple times. They were
surprised WCOM felt having to reboot these boxes all the time was
unacceptable in an environment requiring 5nines availability. Never
laughed so hard when I saw the incredulous faces of the M$ drones. We
brought in a Stratus based solution and won the project.


Alcatel folks where not surprised, I'm sure ;-)
One thing interesting, coming from data background, seeing the 
requirements in carrier voice networks. Is a quite distinct ball-game.
Devices that require 'hot-software-upgrades', still not that often seen 
in data. How is this being handled with Asterisk + other solutions ?
Example, having a trunk gateway with a OC3 worth of TDM, is 'acceptable' 
that a sw upgrade will cut established calls ?


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Re: [Asterisk-Users] ZyXEL P662HW / SIP / Crashing

[Julio Arruda]

Set Wild guess mode on (I'm not familiar with zyxel).:

asterisk-users wrote:


Has anyone experiences this please: -

We were running a number of ZyXEL P662HW-61 routers at our sites and all
traffic was being sent over IP-SEC VPN's between devices.

When we moved to a new architecture, we got rid of the VPN links so that
the SIP traffic was running directly through the routers.  Each site
uses Snom 360 devices with the latest firmware (v4).

 

This means SIP/call signalling and RTP traffic, I assume. Have you 
pinpointed which traffic is the culprit ?
(maybe with routing for the SIP proxy via one router, and for the media 
gateways end points via another ? or SIP via tunnel, rtp outside ?)



Ever since we did this, the routers have been crashing at least 5 times
per day.  They appear to carry out a full cold start each time (as
though they are having a kernel panic).  The ISP is Nildram in the UK,
but we have also experienced this a few times with another router in
France on a France Telecom system.

As soon as we route the SIP traffic via another router, stability
returns to the network.  Our supplier has been very helpful and we have
tested every release of the firmware from the last 8 months, but they
all behave the same once SIP is being transmitted.

The routers are running with their most basic configuration now, but
this doesn't appear to make any difference.
 

Seems like these boxes can do some level of SIP "ALG", looking/messing 
around with the SIP header.
Can you disable this ? You depend in the SIP ALG to handle NAT or 
something like that ?



Does this sound familiar to anyone please?  We are out our wits end and
our supplier has no ideas (and neither do ZyXEL it would appear).

For reference, all traffic is being sent through using G.729, but I
don't think that this makes any difference.
 

If you identify what is the traffic (RTP or SIP), next step would be to 
identify what is the exact trigger (like, if is SIP, is a memory leak in 
INVITEs passing via the ALG and goes on)


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Re: [Asterisk-Users] IAX2 Softphone Quality & Network Cards

[Julio Arruda]

You may want to check if the autonegotiation "agreed" in both sides.
Older nic/drivers/switches would have problems with autonegotiation.

Also, statistics can tell you something about this..
Example, if you have shorts/runts in one port, and late-collisions in 
the L1 'peer' port (the other side of the cable), you may have one side 
in full and the other in half.
(the late-collisions would be counted in the half duplex side, and 
shorts/runts in the full-duplex side)


Adam Robins wrote:

Everything is set to autoneg, NICs, switches and router 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Monday, August 29, 2005 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2 Softphone Quality & Network Cards

Matt Riddell wrote:

 


Adam Robins wrote:


   

Should it be in half duplex or full duplex? 
  

 


Full.

   


AFAIK, depends...
If you have your switches doing autonegotiation, you can't disable
autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL
have a duplex mismatch.
This is as per the standard.
A duplex mismatch is really bad, is in fact worse than having segments
doing halfduplex (properly).
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Re: [Asterisk-Users] IAX2 Softphone Quality & Network Cards

[Julio Arruda]

Matt Riddell wrote:


Adam Robins wrote:
 

Should it be in half duplex or full duplex? 
   


Full.


AFAIK, depends...
If you have your switches doing autonegotiation, you can't disable 
autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL 
have a duplex mismatch.

This is as per the standard.
A duplex mismatch is really bad, is in fact worse than having segments 
doing halfduplex (properly).

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Re: [Asterisk-Users] 911 Notices

[Julio Arruda]

Remarks inline

Dean Collins wrote:


Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.

Lol, how many people do you think took them up on that offer?
 

From what I understand, Packet8 had this option for quite some time. I 
used (more than one year ago) to be Packet8 customer.

I still use a couple of DTA310 in my * system :-)


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Friday, 26 August 2005 6:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 911 Notices

On 8/26/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
   


Broadvoice sent out a notice and threatened to disconnect me if I
 


did
 


not respond. If I disagreed with their stand they would disconnect
 


me
 


too.
   


I think they said something like we don't have it and we ain't
 


getting
 


it. Click here to acknowledge.

I'm guessing that the statement gets them off the hook?


 


The way I understand it. Yes, for now.  That only allows them to
be compliant  up until the mandatory compliance date. After that date
passes, technically, you're supposed to offer it if you're business is
interconnecting voip networks to the PSTN.
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Re: [Asterisk-Users] QoS General Question

[Julio Arruda]

Half duplex by itself doesn't hurt (depends in number of calls and etc 
really, but anyway...)

What is a killer for VOIP is duplex mismatch.
If you have autonegotiation enabled, and your peer (the switch ?) has 
autoneg off, and 100/Full-duplex hard coded, you WILL have a duplex 
mismatch.

And this is as per the spec

Geoff Manning wrote:


Eric Wieling aka ManxPower wrote:

 


In my experience, for local LAN audio issues, duplex problems are the
problem, not LAN traffic.

   



Rock on!

I am in half duplex mode:

serv01:~# ethtool eth0
Settings for eth0:
   Supported ports: [ MII ]
   Supported link modes:   10baseT/Half 10baseT/Full
   100baseT/Half 100baseT/Full
   1000baseT/Half 1000baseT/Full
   Supports auto-negotiation: Yes
   Advertised link modes:  10baseT/Half 10baseT/Full
   100baseT/Half 100baseT/Full
   1000baseT/Half 1000baseT/Full
   Advertised auto-negotiation: Yes
   Speed: 100Mb/s
   Duplex: Half
   Port: Twisted Pair
   PHYAD: 1
   Transceiver: internal
   Auto-negotiation: on
   Supports Wake-on: g
   Wake-on: d
   Current message level: 0x00ff (255)
   Link detected: yes

This could help solve a lot of quality issues.

 



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[Asterisk-Users] Uniden UIP 1868 / Asterisk experiences

[Julio Arruda]

Just wondering..
Any experience with the UIP1868 ?
I assume that it can handle a single SIP line (can't seem to find the 
manual at their site :-)..

They mention also T38 in their webpage.
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Re: [Asterisk-Users] To anyone seeking 911 Service Providers "stay away!!!"

[Julio Arruda]

Andrew Kohlsmith wrote:

On Monday 25 July 2005 23:26, [EMAIL PROTECTED] wrote:


Highly recommended to everyone to stay away from this issue
I do not have a name for the company right off hand, but they got sued
really bad when they tried 911 via VOIP and the 911 drop kept occurring in
different areas and someone died!


From what I understand, was Vonage, and at 
http://www.dslreports.com/forum/voip I remember seeing several posts 
about it..as well as at NANOG mailing list.


Need more data.  Right now you're just fear mongering.  Here in Canada it is 
required by law to have 911 service for your customers.


There where people saying this Vonage issue was all FUD, anyway, side 
effect was...Seems this is 911 for VOIP is "FCC mandatory" now in USA ?
Not sure, I use * at my home and have DSL, so I just route my 911 to the 
landline outbound, I would not expect the outbound IAX providers to 
offer 911 to me :-)



Implement 1 single hard wire for this service and cover your tush!
This is what I tell my customers as well, not because I can't do it but 
because they typically have the line there anyway for fax and/or security 
which carries the DSL circuit for VOIP.  :-)


From what I understand, the big deal is with cable providers and naked 
DSL providers, where you don't really have a 'landline' tied to your DSL  ?
Would be interesting to people involved in asterisk biz. to have a 
summary of countries and their regulations on these issues, anyone know 
if a summary like this is posted somewhere ?


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Re: [Asterisk-Users] IAX over HTTP

[Julio Arruda]

Dave Cotton wrote:

On Fri, 2005-07-22 at 15:42 -0500, Eric Wieling aka ManxPower wrote:


Eric Rees wrote:


We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma.  Most of the time, IAX
sounds better then the land line. 


Using UDP or using TCP?  Might want to confirm by using tcpdump.



OpenVPN uses UDP


OpenVPN can use TCP, and really, I would expect that many users using
openvpn to bypass firewall rules, would be using TCP not UDP.
Simple example...
-
dev tun

proto tcp-server
port 443
-


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Re: [Asterisk-Users] IAX over HTTP

[Julio Arruda]

Eric Rees wrote:

We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma.  Most of the time, IAX
sounds better then the land line. 



Would you have some packet captures ?

Or maybe some tcptrace 'run' over a capture file ?

I would still expect packet loss would be made much worse (side-effects 
wise) in IAX over SSL than in plain UDP/someotherdatagram encapsulation.


Thinking about it, IAX endpoints would not see any packet loss, of 
course, but the jitter would "appear" much worse from the receiver point 
of view, since a packet loss would force the TCP stack to wait for the 
retransmission before sending the "not lost" IAX packets to the 
application, forcing whatever was sent ahead, to be queued in the 
receive side for up to at least RTT (I guess, since the retransmit would 
kick in after at least RTT..humm..too old to remember all TCP fancy stuff)..





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Friday, July 22, 2005 12:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX over HTTP


Just remember that TCP will try to retransmit your lost voice packets,
what is not exactly of any use :-).

VPNs with IPSec and others (CIPE and some UDP 'related' vpns) would not
create this extra (and useless) overhead.
I've used IAX over OpenVPN (with SSL as you), and it does work, to some
level, but I would not do it for a living :-)

Iassen Hristov wrote:


I disagree. Isn't running it over a VPN the same thing?

I have been running with no problems:
a) a soft phone over a OpenVPN VPN (over TCP)
b) a soft phone over a MS PPTP VPN
c) a hard phone over a IPSec net-to-net VPN

For the soft phone I've used X-Ten (SIP) and idefisk (IAX) For the 
hard phone I've used Budgetone BT-102, Sipura SPA-841 and ATCOM AT-320




(w/ IAX2 firmware).

I've had no problems. I suppose it is a matter of a good connection.




Message: 25
Date: Fri, 22 Jul 2005 13:48:09 +0200 (CEST)
From: Jerry Glomph Black <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] IAX over HTTP
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Doing IAX over TCP is simply a Bad Idea.

Under perfect circumstances, it will work OK, but the slightest 
network disturbance will result in sound gaps/distortion and/or 
monster audio delay.


This is not idle UDP-boosting, I've tried it.

[Have had good results with UDP-based secure tunnel transport of IAX 
traffic  (CIPE and OpenVPN)]

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Re: [Asterisk-Users] IAX over HTTP

[Julio Arruda]

Just remember that TCP will try to retransmit your lost voice packets, 
what is not exactly of any use :-).


VPNs with IPSec and others (CIPE and some UDP 'related' vpns) would not 
create this extra (and useless) overhead.
I've used IAX over OpenVPN (with SSL as you), and it does work, to some 
level, but I would not do it for a living :-)


Iassen Hristov wrote:

I disagree. Isn't running it over a VPN the same thing?

I have been running with no problems:
a) a soft phone over a OpenVPN VPN (over TCP)
b) a soft phone over a MS PPTP VPN
c) a hard phone over a IPSec net-to-net VPN

For the soft phone I've used X-Ten (SIP) and idefisk (IAX)
For the hard phone I've used Budgetone BT-102, Sipura SPA-841 and ATCOM
AT-320 (w/ IAX2 firmware).

I've had no problems. I suppose it is a matter of a good connection.



Message: 25
Date: Fri, 22 Jul 2005 13:48:09 +0200 (CEST)
From: Jerry Glomph Black <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] IAX over HTTP
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Doing IAX over TCP is simply a Bad Idea.

Under perfect circumstances, it will work OK, but the slightest network 
disturbance will result in sound gaps/distortion and/or monster audio

delay.

This is not idle UDP-boosting, I've tried it.

[Have had good results with UDP-based secure tunnel transport of IAX
traffic  (CIPE and OpenVPN)]


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Re: [Asterisk-Users] IAX over HTTP

[Julio Arruda]

Not that it need any additional 'push' against it, :-)..

My tests with IAX over OPENVPN (on port 443) are acceptable (they do 
work just fine) for basic non-user-friendly purposes.


Examples, I get my voice mail at home sometimes via this tunnel (if wife 
using primary landline.

I test my dialplan via firefly over this tunnel.
I called my family over FWD but really nothing to be used for anything 
that really matter. Would not pass any "non-geek" acceptance test.



Jerry Glomph Black wrote:

Doing IAX over TCP is simply a Bad Idea.

Under perfect circumstances, it will work OK, but the slightest network 
disturbance will result in sound gaps/distortion and/or monster audio 
delay.


This is not idle UDP-boosting, I've tried it.

[Have had good results with UDP-based secure tunnel transport of IAX 
traffic (CIPE and OpenVPN)]





On Fri, 22 Jul 2005, Tzafrir Cohen wrote:


On Fri, Jul 22, 2005 at 11:39:04AM +0200, Gustavo García wrote:

You can traversal a HTTPS proxy using a plain TCP connection (without 
SSL).
The unique requirement of some HTTPS proxys is that the target port 
is 443.


Then if your Asterisk listen in 443 port IAX (TCP) connections, it 
should

work.



which makes this wishlist item a simple dependency of the wishlist item
for IAX over TCP.


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Re: [Asterisk-Users] IAX over HTTP

[Julio Arruda]

Rob Scott wrote:

For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing IAX over HTTP?

I have heard of projects such as stunnel.

Has anyone tried something like this?

I did a quick search but didn't come up with much.


I did some tests, with openvpn, for my purpose, was ok, not sure how 
would behave in packet loss, jitter conditions..

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Re: [Asterisk-Users] How to avoid collect calls on ISDN BRI trunks?

[Julio Arruda]

Dhennys,
I would expect that the ISDN collect call would have some kind of 
notification about the charge.
In E1/R2, the Telebras standard in fact DOES have this notification 
defined, from what I remember, the problem was that many of the CO 
switches would not support it, that is why the 'double-answering' is 
used in most if not all the cases (there is a condition where the 
double-answering would tear down any call, something to do with the 
configuration in the PSTN switch).
I don't think you can answer/hangup/answer a call in ISDN the way you 
can in E1/R2 or analog lines...



Armin Schindler wrote:

On Tue, 12 Jul 2005, Dhennys Pestana wrote:


I need to block collect calls on my PBX.

I was able to find information on Google regarding ISDN ZAP channels, but not
ISDN CAPI channels which is my case.

Since there's no information from the Telco that the call is going to be charged
by the callee, if a particular call is automatically answered by the PBX instead
of a real person there's no way to avoid it. There's only a recording AFTER the
call is answered, asking for the callee party to accept it.

By using ZAP channels with HFC ISDN cards, it's possible to use Hangup() and
Flash() commands, which won't do any good on CAPI channels because it will
hangup the call immediatelly.

Note: This situation is exctaly the same with remote access servers (RAS),
commonly used by Internet Service Providers.

Here's an example of what should happen on my scenario:

[default] ; External calls comes on "default" context
exten => s,1,Wait,1   ; Wait for all ISDN and CAPI messages
exten => s,2,Answer() ; PBX actually answers the call
exten => s,3,Wait,1   ; Just in case
exten => s,4,Flash()  ; Avoid collect calls, don't actually hangup
exten => s,5,Wait,1   ; Wait a second (just in case)
exten => s,6,Answer() ; Now it should work as if it were "s,1"
exten => s,7,BackGround(IVR-menu) ; 1 for sales, 2 for support, 3 for...


Note: Step "s,4" could be also "Hangup()", it won't make any difference.

If it were an analog trunk, it would work flawlessly. Unfortunatelly for digital
trunks, when you hangup the channel the call is dropped immediately.



Yes, because Hangup() means 'disconnect'.
 
I'm not aware of Flash() and what it is doing, but it is surely not 
implemented in chan_capi.
If someone can tell me what Flash() is supposed to do, we can implement it 
in chan_capi-cm.



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Re: [Asterisk-Users] IAX DTMF Problem...

[Julio Arruda]

Denis Galvão - iSolve wrote:

IAX doesn't use INBAND DTMF.

Denis Galvão.


Denis,
A clarification, I hope, just to make Mark aware of the small difference.
IAX sends DTMF in the signaling 'stream', that happens to follow the 
same path as the media.
But, in IAX DTMF is not sent as voice payload (as the typical inband 
when is being used in SIP as one example), just happen to be sent "in 
the same flow" from source host/port to destination host/port.
In SIP, when we use "inband", in fact, the DTMF tone will be treated as 
a voice 'piece', and will be sent in RTP packets without any special 
considerations :-)
From what I understand, that is one of the reasons with SIP inband 
doesn't mix well with any codec other than G.711.

(compression could distort the tone).
Mark, you may want to ask your provider to check the same kind of trace 
you did in your side.
(after all, it seems much like you ARE sending the DTMF to them, is not 
like will get distorted :-)




On 01 de jul de 2005, at 03:23, Mark Edwards wrote:




Hi.

Probably been asked before, but my IAX provider assures me its not  
their problem


I have a IAX connection to a peer providing a DID. I am dialing up  my 
number, seeing the DTMF tones come down the line, and the * IVR  is 
just ignoring them.


IAX debug output is:

Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1
   Timestamp: 02608ms  SCall: 00016  DCall: 3  [ 210.80.176.12:4569]
Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX  
Subclass: ACK

   Timestamp: 02608ms  SCall: 3  DCall: 00016 [210.80.176.12:4569]

for a press of "1"

I am assuming this is the DTMF inband problem, but I appear unable  to 
convince my provider.


Can I work around this on * or do I have to go back to SIP?


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Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

[Julio Arruda]

[EMAIL PROTECTED] wrote:

2- Out-of-band is as safe/unsafe as having the conversation recorded,
including pin, by the hacker, if no encrypted voice path is being used.
as others mentioned, DTMF tones would be very "obvious" in a trace
(maybe someone may want to post an example).



Watch out:

http://www.asteriskbrasil.org/tiki/tiki-browse_image.php?imageId=17



Denis, I'm sure RFC2833 (or INFO for all that matters) would be easy, 
what I mentioned, is that, with inband, it for sure still easy to 
capture it, after all, the other end-point NEED to be able to regenerate 
the DTMF back into the PSTN, so, best case, inband DTMF is security 
through obscurity (and being G.729 and the other codecs a well known 
standard, building a "DTMF recovery ethereal plugin" should not be 
exactly rocket science..)

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Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

[Julio Arruda]

Andrew Kohlsmith wrote:

On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:


But there are some products that supports DTMF inband on G729. Ok, it will
not work in most cases(like everyone told) but why Asterisk dont support
it? Is this hardcoded, or is possible to try it out?



Asterisk can do it too, it's just not reliable on any platform.  Set 
dtmfmode=inband and use the g729 codec; that's all there is to it.  You will 
be disappointed though.


I think that IAX, as one example, won't allow this ? Have a faint memory 
of some error message when trying it (maybe was ILBC+iax ?)

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Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

[Julio Arruda]

[EMAIL PROTECTED] wrote:

On Thursday 23 June 2005 19:57, Brian West wrote:


With inband its at least not sent in clear text.


It's trivial to pull DTMF out of an inband stream too.  Perhaps not AS
trivial
but just the same, you should be using SRTP if you're paranoid about this
kind of thing.



We are on a real world... Every cyber cafe has its own little
hacker/cracker that is sniffing out... A simple ethereal capture could
give me a bank pin number... It is REALLY trivial!


I think the point(s) the others are trying to make:

1- It is not feasible to use inband in G.729 (or, as far as I know, any 
other compressed codec), and that is final. Other than that.


2- Out-of-band is as safe/unsafe as having the conversation recorded, 
including pin, by the hacker, if no encrypted voice path is being used. 
as others mentioned, DTMF tones would be very "obvious" in a trace 
(maybe someone may want to post an example). Remember, if the other end 
need to be able to "regenerate" the DTMF info, it MUST be present in the 
stream, so is as easy/hard as the other endpoint 'decoding' it.


PS: I seem to recall some Voice over data products that would upspeed to 
G.711, upon detecting of DTMF tones, this may have given someone the 
wrong impression, that the DTMF was being sent as G.729, when it was not 
in fact.

[], 




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Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?

[Julio Arruda]

Roman Zhovtulya wrote:

Dear all,
I've noticed some significant voice quality deterioration when calling US
landline via VoIPjet.com in the last week or so.
Before that the quality was pretty good.
Has anyone else experienced any voice quality problems with voipjet
recently?


I've been using VOIPJET for Brazil LD without any problems.
(or should I say, my wife has been using, still can't thank VOIP enough 
for the savings..)

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Re: [Asterisk-Users] new cisco ip video phone?

[Julio Arruda]

Anyone tried the Packet8 Videophones ?
I would guess that leadtek is providing the "non-branded" version now ?
[], 

Dean Collins wrote:

I've played with the dlink eyebeam but only for ip to ip calling not
used with asterisk.
It's crap.


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Thursday, 26 May 2005 3:12 PM
To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] new cisco ip video phone?

I noticed several obviously fake screens on '24' and passing reference


in


the script to "intrusion detection systems" by Cloe, all with the


Cisco


logo
featured prominently; this is undoubtedly product placement payola. I
suppose it's possible that Cisco is deploying some preproduction sets


to


the
'24' set to build some hype in the '24' demographic, but occam's razor
suggests that this is just some cisco phone shells with a standard LCD
inside slaved to played back video or a videocamera at the other end,


or


maybe even inserted digitally post-production.

I'd like to hear if anyone on the list has tried the Dlink EyeBeam


thingy.


At least, *that's* a product that is shipping.

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Re: [Asterisk-Users] Who knows where voicepulse has their asterisk servers?

[Julio Arruda]

InternetMarketingMan2001 wrote:
I want to collocate an * box somewhere, where better than where voicepulse
chose to put their servers?
They probably did their homework and selected someplace where good handoff
to the pstn can be found, right/
AFAIK, the voice path doesn't really need to follow the same path as the 
call signaling.
I assume they could have the media gateways really far from the Asterisk 
servers, would this make any difference in your plans ?
Anyone checked the source/destination for traffic going via voicepulse ?
For several destination DNs ?

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Re: [Asterisk-Users] Voip Provider in Brazil

[Julio Arruda]

Asterisk wrote:
Hi all,
Is there a VOIP provider that can deliver local Rio de Janeiro numbers?
I am looking for a normal Rio number for my Asterisk box.
I'm using a RJ DID, right now http://www.libretel.com with IAX DID (they 
offer SP also).
Have not tried much on it, noticed DTMF can be a little picky, butdidn't 
try anything on troubleshooting.
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Re: [Asterisk-Users] Grandstream firmware 1.0.6.2

[Julio Arruda]

Doug Lytle wrote:
Grandstream owners,
I just noticed that there is a new firmware release, for those that are 
interested:

http://www.grandstream.com/BETATEST/
2 quick notes, a quick test seem to indicate iLBC is broken (didn't try 
any troubleshooting).
And, in the release notes, from what I remember, there are mentions of 
problems with dowgrading it, at least they recomend you to call support 
to do it)
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Re: [Asterisk-Users] Channel bank of E1s? (one E1 input --> 2 x E1 output)

[Julio Arruda]

Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.

Matteo, would you have any reference for this 'mux/splitter' ?
I would guess it need to be smart enough to dig into the signalling, 
since is not only the PCM DS0s that would need to be "Y-splitted".
[], 

Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
Hi,
Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs ,
one with 15 channels and the other with 15 channels;
Is there a sort of E1 multiplexer devise that allows me to plug in one hand the
E1 port of the Digium card and on the other hand the two PABXs? In this same
devise, I should be able to say that 15 channels need to go to first Interface
and 15 other channels need to go to other interface.
Or is it necessary to acquire a another E1 card although I don't need to process
more channels (30 channels are ok).
Any help is greatly appreciated.

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Re: [Asterisk-Users] IAX aproprietary protocol

[Julio Arruda]

Stefan de Konink wrote:
On Wed, 27 Apr 2005, Joseph wrote:
How can proprietary protocol be open protocol?

If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Digium, thus proprietary.
http://en.wikipedia.org/wiki/Proprietary
But there are royalties or something like that ?
I understand that proprietary protocols CAN be published, but what make 
them proprietary is the requiremenf or royalties or at least a 'ok' from 
the owner ?
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Re: [Asterisk-Users] Re: Cisco to buy Sipura

[Julio Arruda]

Tom Ivar Helbekkmo wrote:
Leo Ann Boon <[EMAIL PROTECTED]> writes:

My prediction: 2 years down the road, they'll leave again and set up 
SipZilla to make another low-cost ATA to compete with the SPA-.

...and then they'll sell *that* to Cisco, too.  :-)
Or, 2 years down the road, VOIP will be so usual, that ATAs won't be 
that popular (in the end, it can be seem just a 'to connect to legacy 
device' adapter).
Maybe lower cost IP phones (wifi or ethernet) at homes will be the rule 
? :-)

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Re: [Asterisk-Users] VOIP Regulations in INDIA

[Julio Arruda]

Vikram Rangnekar wrote:
Does anyone know the exact VOIP regulations in India. What I want to know is
that are VOIP EPBX with E1 lines allowed for comapnies in India. For example
If I am a company which has 1 incomming E1 line can I have SIP or IAX
extensions inside my office and receive that PSTN call on the VOIP extension. 

Another senario would be say my office has a branch in another state and I
have a leased line or some sort of data network between the two offices can I
tranfer a PSTN call comming in over the E1 in one state to a VOIP extension
in another state over the data link between the two offices. 
Telecom in India is over regulated and according to what I figured out that
both those senario's above are not allowed I hope someone can prove me wrong.
After reading 
http://www.investindiatelecom.com/BPO%20Regulations/Domestic%20-outgoing.htm
and
http://www.dotindia.com/isp/guidelines.doc

I dont see much hope for offices in India who want to use Asterisk with say
Digium E1 cards as they would need to give a bank draf of 10 lakhs to DOT to
use VOIP internally. 
Vikram,
You may want to contact the local India entity responsible for the 
legislation.
There are many details in some countries on the specifics. The other 
thing you may need to be aware is that some countries may be now (or in 
the future) filtering VOIP traffic (search in google), so you may need 
to check if the Internet provider you would be dealing with (you mention 
leased line, that should not be a problem so, but just being generic 
here..).
I know some other countries do have specific constraints (Panama ? 
Mexico ? Brazil ?) and each one a little distinct from the other..

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Re: [Asterisk-Users] Re: asterisk@home scary log

[Julio Arruda]

One good step is to 'test' your public IPs against any mistake/hole like 
this.
I've used http://www.ordb.org in the past for this purpose, others for 
sure are available.
I would assume is a valuable feedback to provide to the folks from 
[EMAIL PROTECTED], to have a more "conservative" configuration in their 
default install.

Jean-Louis curty wrote:
hummm if that's the case I might not be the only one!
I only installed the [EMAIL PROTECTED] iso (based on centos distro )and
did not change a little comma of the configuration of sendmail,
MTA is configured by default already by [EMAIL PROTECTED]
>
On Thu, 10 Feb 2005 11:09:29 -0500, Jason Stewart <[EMAIL PROTECTED]> wrote:
On 10/02/05 15:10 +0100, Jean-Louis curty wrote:
so I stopped asterisk, type mail and got a strange mail saying that
user [EMAIL PROTECTED] could not be reached and body was like if it was
the result of commands ifconfig etc
unfortunally I don't have the message anymore but I went to the log
Feb  9 20:30:17 asterisk1 sendmail[10093]: j1A1U7mf010088:
to=<[EMAIL PROTECTED]>, ctladdr=<[EMAIL PROTECTED]> (0/0),
delay=00:00:10, xdelay=00:00:10, mailer=esmtp, pri=30329,
relay=gsmtp171.google.com. [64.233.171.27], dsn=2.0.0, stat=Sent (OK
1107998984)
the thing is i did not send any message to [EMAIL PROTECTED] nor to
somebody at yahoo,
anybody got the same ? what can I do ??
There's a chance that you may have been hacked, but the logs you post
look more like your mailserver is an open relay. What OS/Distro are
you using, what version, and do you have the latest patches applied?
What services are you running?
Look for strange entries with uid 0 in your passwd file. Also check
for root kits with a rootkit checker (chkrootkit.org).
If everything pans out security-wise then the only problem is that you
MTA is configured to be an open relay. If that's the case, then you
need to fix it right away before you get on umpteen million blackhole
lists.
Consult the docs and/or community for the MTA that you're using to fix
that.
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Re: [Asterisk-Users] Cisco IP Phones

[Julio Arruda]

Keith Burns wrote:
I think you need to look at a few other factors.
...
2. Line power - Cisco uses one standard, other phones use another... but
Cisco is the 900# gorilla in the powered switch market... your call...
I'm curious about this point..
Most if not all vendors that support PoE are not already support 802.3af 
standard ?
...
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Re: [Asterisk-Users] PIX!!!!!

[Julio Arruda]

Christopher wrote:
Can anyone point me in a good direction for configuring SIP through a 
PIX using 1:1 NAT.  I have read anything I could get my hands on and 
tried them all with very little success.  I can get it to work through 
the cheap little cable modem routers, but not this PIX.
I -can- make a direct SIP call using the IP address of the * server 
([EMAIL PROTECTED]), but when I do that * still doesn't show it 
registering.  Even when I call through this method the phone comes up as 
"UNREACHABLE" and the port is listed as 0 instead of 5060 like all the 
internal phones.
I seem to recall some weird thing in the PIX, where you had to disable 
the SIP fixup to work (and of course, to use some nat traversal trick, 
like outbound proxy).
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Re: [Asterisk-Users] QOS / Cisco / Asterisk

[Julio Arruda]

Matt Schulte wrote:
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We were
trying to match by TOS set by Asterisk however it seems we've run into a
snag where the packet TOS tends to get reset somewhere on our network.
Has anyone had this issue? We're running Cisco everywhere inbetween
(even the switches). Is there an alternative way to match these? We've
thought of by port but that's kind of ad-hoc IMHO.
I know some LAN switching devices, in a default "QoS" configuration, 
would treat ports as "diffserv" untrusted ports, or access ports, 
meaning, the DSCP (a reuse of the TOS also) in packets inbound at that 
port are not to be trusted. Have you looked at your switches documentation ?

Asterisk1 --> 3560 --> 2600 -- (T1) --> 7500 --> 2900 --> 3550 -->
Asterisk2 

Sniff: (note the dumps between the 2 machines are diff times however
they show the same occurance)
Asterisk1: 1.1.1.1
09:09:10.019191 IP (tos 0x10, ttl  64, id 58, offset 0, flags [DF],
proto 17, length: 60) 1.1.1.1.12056 > 1.1.1.2.19726: [no cksum] UDP,
length 32
09:09:10.030146 IP (tos 0x0, ttl  62, id 63, offset 0, flags [DF], proto
17, length: 60) 1.1.1.2.19726 > 1.1.1.1.12056: [no cksum] UDP, length 32
Asterisk2: Dump on 206.80.70.55
09:34:34.418386 IP (tos 0x0, ttl  62, id 261, offset 0, flags [DF],
proto 17, length: 60) 1.1.1.1.14796 > 1.1.1.2.18996: [no cksum] UDP,
length 32
09:34:34.422974 IP (tos 0x10, ttl  64, id 273, offset 0, flags [DF],
proto 17, length: 60) 1.1.1.2.18996 > 1.1.1.1.14796: [no cksum] UDP,
length 32
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Re: [Asterisk-Users] Free World Dialup and Asterisk

[Julio Arruda]

Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX 
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby 
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in dialup, 
correct? From what I recall, FWD would do only G.711, would not exactly 
work in dialup (maybe ISDN with 2 b-channels ?)
PS: I don't see the dialplan for the "inbound" calls, where a call from 
FWD would land in your * ?

Gonzalo Gasca Meza wrote:
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
 
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.

 
  Spain LAN
FWD dialup account -> Internet <-- 3COM router/switch --- Asterisk -- 7960
 
I have done some research in google with no success.
http://www.m-networks.net/home/asterisk/ast-fwd.htm
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
 
 
When I connect my FWD client in the LAN i can dial FWD numbers
ANY IDEAS OR CONF FILES WORKING WILL BE APPRECIATED
THANKS!
 
 
 
 
 
server*CLI> sip show registry
Host  Username   Refresh State
69.90.155.70:5060 431044 160 Registered
69.90.155.70:5060 421058 160 Registered

 
SIP.conf
register => 421058:[EMAIL PROTECTED]/103 ;Register Free World Dialup
register => 431044:[EMAIL PROTECTED]/103
[fwd1]
type=friend
username=431044
secret=password
fromuser=431044
fromdomain=fwd.pulver.com
host=fwd.pulver.com
insecure=very
canrenvite=no
nat = yes
dtmfmode=inband
 
[fwd2]
type=friend
secret=password
username=421058
fromuser=421058
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no

extensions.conf
FWDUSERID1=421058
FWD1USERNAME=Gonzalo Gasca
FWDUSERID2=431044
FWD2USERNAME=Gonzalo Gasca
FWDPREFIX=*
[fwd1-out]
exten => _8.,1,SetCallerID(${FWDUSERID2})
exten => _8.,2,SetCIDName(${FWD2USERNAME})
exten => _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
 
[fwd2-out]
exten => _7.,1,SetCallerID(${FWDUSERID1})
exten => _7.,2,SetCIDName(${FWD1USERNAME})
exten => _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _7.,4,Macro(fastbusy)
exten => _7.,5,Hangup

My IP phone include those fwd1-fwd2-out
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Re: [Asterisk-Users] VoIP Termination

[Julio Arruda]

Mike Diehl (Encrypted email preferred) wrote:

I'm trying to make an issue out of this because I think it needs to change and 
I'm hoping people who are affiliated with these providers are reading this.  
I was going to go with Packet8.  I was going through the "final checklist" 
before subscribing when I came accross this fascist policy.

Sure, I can go with a business plan, but that would cost me $39.95.  That's $5 
more than I'm spending for an analog phone line!  Part of the reason for me 
to go with VoIP is to become "Quest Free."  But suddenly, Quest is starting 
to resemble the Boy Scouts when compared to the types of usage policies I'm 
seeing from some of the VoIP providers.
Sorry for the rant, but I hope you understand.

Mike,
I was a happy P8 user for 18 months, just keep in mind that theirs (and 
most of the "consumer market" voip providers) unlimited plans could be 
in theory abused by telemarketers and etc.
I just cancelled P8 because their rates to Campinas/Brazil went too 
high, but is after all, consumer VOIP.
You will notice that almost all the "unlimited VOIP plans" have some 
kind of safeguard/policy like that. The good part is, IMHO, with VOIP, 
you are really free to choose :-) and running asterisk just give you 
even more flexibility.
You may want to take also a look at http://www.dslreports.com/forum/voip
They have quite few users of each of the consumer VOIP providers, and 
you should be able to have a feeling about how good/bad are their 
experiences.
[], 

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Re: [Asterisk-Users] Voipjet problems

[Julio Arruda]

I would guess they really had some problems yesterday.
I had some failed calls in my * at home, had it rerouted via Nufone, 
since the mother-in-law-retry timer was set too low, and I didn't want 
to hear complaints when I arrived home :-). I'll try to switch it back 
later today.

Ed Greenberg wrote:
I made a few voipjet calls today and they all went through just fine.
--On Wednesday, December 15, 2004 9:26 PM -0300 Gustavo Russo 
<[EMAIL PROTECTED]> wrote:

 Anybody is experimenting problems with Voipjet lately ?
Last 2 days we are having some intermitent problems in which, after
accepting the call, the error at the Asterisk console is :
 == No one is available to answer at this time
 Voipjet technical support at this time was not able to fix the problem.
 Regards
Gustavo
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Re: [Asterisk-Users] How can i test a modem with Asterisk?

[Julio Arruda]

Fabrício Zimmerer Murta wrote:
Oh, friend... I have realised just yesterday that's impossible to use
regular modems (say hayes/v90 33.6 or 56k) to plug asterisk to the world. I
can't figure out why. But they simply don't support it.
  If you want to use your isdn modem to plug * to the world, it's OK.
Else... Only just one modem kind is supported and, surprisingly it's a
winmodem, from Intel. It's Intel 537 w/ chipset MD32000. That's the sole
only modem supported to be used as a * node to link to the PSTN (telco) and
start to make calls from within the VoIP network (or whatever that passes
throught asterisk).
  I'm thinking in sending a mail for asking WHY THE HELL they can't support
bare modems, even if they have voice support (I have an USR w/ voice, 56k
and ISA kind, and I simply can't use it for testing an * box).
  Also, I think that with that computer you can gladly test you *
implementation as long as you don't use much concurrent connections.
- Fabrício
Google is your friend, as well as the wiki, don't leave home without it 
: :-)

http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware
As someone else mentioned, the winmodem will pass all the info to the 
host CPU, while a "regular" modem will not.

In .br you will find vendors for these cards, from what I understand, 
few have pointed even some stores (SP, St Efigenia) that sell these.
From what I can see in the asterisk-br, just found a RJ address:

"LE Compustore
Placa Modem Intel com chip MD3200
Procurar : Aline Compustore 21 2262-7169
Av. Rio Branco 156 Loja 301"
On the Why the heck :-) part, I guess that you can try to fund any other 
development you may need, keep in mind that, while the "Hayes/V90, USR 
w/voice" and etc would not work, maybe others winmodems can have a 
driver developed ?
I've tested (and keep using at home) the Sipura 3000, seems to work well 
enough, and I may even get rid of my FXO card, just need to understand 
clearly the timing gotcha x conference (the FXO does timing to merge the 
audio streams AFAIK).


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Re: [Asterisk-Users] Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk?

[Julio Arruda]

Jerry Glomph Black wrote:
I have a lot of experience, all of it pretty good, with various Sipura 
products, Grandstreams,  Zultys, IAXy, and numerous SIP/IAX soft phones 
connecting into Asterisk as clients.   Good sound quality, great 
reliability.

I've tried two of the units named in the subject line, and frankly I'm 
frustrated.   Calls usually start out OK, but within a brief period the 
sound goes totally to Hell.  Sounds like the packets are being 
reassembled out of order, because there is a regular candence to the 
garbling.  Problem is almost always on the receiving end, the distant 
party on the call seems to get OK audio.

Most annoying is that when I log the device directly into a VoIP 
provider (have tried FWD, Stanaphone, and Sipgate.de) IT WORKS FINE!

I've tried asterisk boxes on the local LAN, and thousands of miles away. 
Asterisk versions from 0.7.2  to 1.0.3.

Results have been consistently flaky, I've tried flash upgrading, makes 
no difference.   Have tried all sorts of config tweaks on the phone as 
to buffer size, etc.

Google has almost NO info on these things, they have one nice feature 
which is easy autoswitching between POTS and SIP calls in both directions.
Any experience or hearsay out there in Asterisk land?
Not very helpful, I know..but..
From what I understand, these are based in the same hw/sw as the 
Packet8 DTA310 ? (audacity based gear)
I use DTA310 for some time with * and seems to work fine for my 
purposes, and with good quality.
Anyway, you may want to post the configuration you use for the leadtek, 
may give the others a hint ?
[], 

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Re: [Asterisk-Users] Unable to create channel of type 'Zap' (cause 0)

[Julio Arruda]

Remarks inline
Leonardo J. Tramontina wrote:
No, I'm not in USA!!
Besides this, my Asterisk is not making external calls; it is installed 
for some tests...

I forgot to say... I'm making these tests in order to test the TE110P 
card we bought.

- Original Message - From: "Steven Critchfield" 
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Friday, December 03, 2004 6:14 PM
Subject: Re: [Asterisk-Users] Unable to create channel of type 'Zap' 
(cause 0)


On Fri, 2004-12-03 at 16:39 -0300, Leonardo Tramontina wrote:
Hi,
I've created a test at "extensions.conf" like this with 3 steps:
; When dial , get the first available channel and dial do 482343400
exten => ,1,Dial(Zap/g1/482343400,5,rt)
482-343-400
Are you missing a number? Looks like it to me, well if you are in the
US.
Brazil, a DN has 8 digits. This way for quite some time.
This would be a local dialed call, for a regular phone.
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Re: [Asterisk-Users] Sveasoft Alchemy QOS

[Julio Arruda]

Steve Kennedy wrote:
On Wed, Dec 01, 2004 at 02:53:50PM -0500, Kanuri, Seshu (Company IT) wrote:

Tell me which one can get me access to the LinkSys Linux using SSH? Does
Satori has this feature? I am not so concerned with Voice Shaping and
QOS at this time, but more interested in converting this into a Linux
box that is accessible from an ssh client.

Alchemy has ssh access, you need to pay $20 subscription to Sveasoft to
access the pre-release firmware.
Steve
I've used satori with SSH (I do have a subscription, so I've a mix of 
satori and alchemy right now, 2 x wrt54gs). SSH Works just fine in both.
There is one feature that I miss, is the option to do the QoS based in 
DSCP markings, most VOIP devices do a clean job of doing the marking 
already, so for outbound QoS is easier than going around with 
filters/port ranges and etc..I guess I'll submit a feature request :-)
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Re: [Asterisk-Users] Packet8 integration into Asterisk?

[Julio Arruda]

dean collins wrote:
Hi John,
I've been using Packet8 via a physical ATA and XP100 card for some time.
As far as I know it is not possible to connect to the Packet8 service
without the ATA.
If this is not the case I would be very interested to hear this.
In addition since moving to the USA I now only have a single packet8
line into my asterisk box (I used to have this and a 2nd regular pstn
line)
I used to be able to forward calls from packet 8 box out the pstn to my
mobile when away from the house but with a single line this
configuration is no longer available.
Does anyone know how to utilize the 3 way conferencing feature of
Packet8 to enable a call to come into my asterisk box and then back out
to my mobile (or whatever forwarding number I allocate).
Basically I need the asterisk server to answer the incoming call, put
the call on hold, dial my mobile number once answered connect the
original incoming call to my mobile.
Not sure if this helps...
You may want to play with call forwarding to a FWD number..
I used to have this setup, P8 -- FXO interface into an asterisk box.
The call forwarding from Packet8 to FWD allowed me to receive all calls 
directly into the Asterisk bypassing the DTA (it would ring once from 
what I recall, but I had a Wait on this FXO port to give the "FWD" leg 
of the call to be answered by one of the SIP phones).
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Re: [Asterisk-Users] Cisco ATA and G729

[Julio Arruda]

Brian Wilkins wrote:
You can only use g729 in pass-thru mode without paying for the licensing fees. 
G729 is probably the best codec around. If you plan on having any sort of 
thriving business based on VoIP, g729 would be the way to go. I don't suggest 
PCMU or PCMA for production. The ATA will pass a list of supported codecs to 
the Asterisk server and based on what you have allowed in your h323.conf or 
sip.conf file, that will be what codec is selected. Your audio quality 
problems could also be traced to a problem with transcoding between different 
codecs (i.e alaw -> ulaw problem). I suggest you try one by one, all the 
codecs available to you and disallow/allow codecs in your configuration until 
you can find the source of your problem. 

Humm..How well is G.729 with Music on Hold ?
I've used G.711 for some time (and now, some iLBC), but of course, this 
is a home system, and need to pass the "wife test". I would even go 
G.722 if required :-)
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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Steve Underwood wrote:
Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. 
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. 
I know
not to use 729 for faxing. Which 'should' I use?
...

Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?

I would expect FAX over ISDN to work better than fax over G.711 on 
some networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is 
a circuit switched technology.

Clue: G.711 is the codec used over ISDN :-)
Ok, now I understand why you mentioned that common codec, let me clarify 
my remark (I can blame on english not being my native language I guess 
:-)..,
'replace' fax over G.711 by fax over G.711 over IP :-)
The point being, Fax over VOIP (even using G.711), I don't believe would 
be as reliable as Fax over an ISDN b-channel :-) Better now ?

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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Steve Underwood wrote:
Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. 
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. 
I know
not to use 729 for faxing. Which 'should' I use?
...

Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?

I would expect FAX over ISDN to work better than fax over G.711 on 
some networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is 
a circuit switched technology.

Clue: G.711 is the codec used over ISDN :-)
I understand a voice switched PCM channel/DS0 is G.711 and this also is 
the b-channel 'unrestricted 64k' ISDN circuit.
What I'm saying is, with ISDN, you have a end-to-end circuit switched 
channel, with very low delay and jitter, with g.711 over VOIP, you have 
at least the additional sampling time (20ms ?), jitter buffers, variable 
delays in the routers in the path (queueing related).

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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. I 
know
not to use 729 for faxing. Which 'should' I use?
...
Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?
I would expect FAX over ISDN to work better than fax over G.711 on some 
networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is a 
circuit switched technology.


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Re: [Asterisk-Users] Re: UDP Fragmentation Problem

[Julio Arruda]

Tom Ivar Helbekkmo wrote:
Bastian Schern wrote:

How I can setup Linux to handle UDP fragments?
There's no setting up to do -- it simply handles them correctly.  Any
IP stack has to.  The problem isn't there, but is an unfortunate
interaction between the sender and gateway/firewalls along the way.
Julio Arruda wrote:
| Butit is quite weird they have such a small MTU. Many websites
| that have problems with Path MTU discovery would be broken by that
| (dumb websites, but still, way too many...).
The web sites in question aren't the problem.  Again, it's things that
happen to the packets along the way that cause communication to fail.
..
Modern Microsoft IP stacks do path MTU discovery by default, which
means that the problem is often seen when accessing IIS web sites.
But it's not "dumb websites", it's dumb firewall administrators. :-)
Humm...I've to disagree..Let me rephrase, I do agree with the despise 
for the blocking madness, and this is the actual cause for breaking 
pmtu, BUT, IMHO, a decent TCP stack should BY DEFAULT detect the PMTU 
Discovery blackhole, and fallback accordingly. That was not the default 
in MS from what I understand, maybe the fixed it ?
Stealing from someone else "be conservative in its sending behavior, and 
liberal in its receiving".

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Re: [Asterisk-Users] UDP Fragmentation Problem

[Julio Arruda]

Bastian Schern wrote:
Hi everybody,
I've got no success to get a friend in Bogota (Colombia) connected to my 
Asterisk. He has got a ISDN Internet connection and the UDP packets will 
be fragmented. It seems that the MTU of this connection is round about 
400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented.
Is Asterisk not able to handle fragmented UDP packages?
Is it possible to use SIP over TCP with X-Lite?
Or has somebody another hint for me?

Fragmentation should not matter for the end-point (the source or 
destination of the UDP datagram), since the IP stack itself should take 
care of the reassembly..
Butit is quite weird they have such a small MTU. Many websites that 
have problems with Path MTU discovery would be broken by that (dumb 
websites, but still, way too many...).
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Re: [Asterisk-Users] Nortel Phones.

[Julio Arruda]

Remarks inline:
Cian O'Sullivan wrote:
Hello,
I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on
their asterisk implementation.  My local dealer says they are not
compatible with any open source implementations.  Is there a phone
compatibility list somewhere?
First, 2 disclaimers...I work for Nortel and I don't speak for Nortel 
(huh ?)...
The I2004 for sure (the original one, I understand there are distinct 
versions) would run just a protocol known as Unistim (you may say is 
like the Skinny protocll for a cisco phone).

[], 
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Re: [Asterisk-Users] Direct SIP connection to Vonage service

[Julio Arruda]

[EMAIL PROTECTED] wrote:
Do you have a list of those providers that use IAX?
http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers
is a good starting point..
Try a search on google, you would be surprised on how many of these will 
pop...


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
on Asterisk Mailing Lists
Sent: Friday, October 22, 2004 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Direct SIP connection to Vonage service
On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson <[EMAIL PROTECTED]>
wrote:
I would appreciate your opinions on the feasibility of these
techniques, and also about any other methods that have been
tried to achieve direct SIP connectivity.

If you are that desperate to use Vonage, then why don't you sign up
for the secondary soft-client option which is $15 or so IIRC?! That
will allow you to connect Asterisk directly to Vonage, although you
pay extra for the privilege.
I personally wouldn't bother and I wouldn't want to take my money to a
company that uses a business model that I despise. So, vote with your
wallet. Don't use Vonage. Use a true VoIP service. And while we are at
it, support IAX: Use a provider that offers IAX.
rgds
benjk
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Re: [Asterisk-Users] DTMF G729

[Julio Arruda]

JOAO CARLOS MOURA wrote:
Thank you Michael,
I tried to use RFC2838 without success. Which another type?

Which endpoints (SIP Phones, ATA, ???) are we talking about ?
You need to match the configuration on the end-point, it may seem 
obvious, but if you leave your IP phone doing dtmf inband, while the 
Asterisk config say it expect it as INFO or RFC2833, it won't work.

- Original Message - From: "Michael Bielicki" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Friday, October 22, 2004 6:14 PM
Subject: Re: [Asterisk-Users] DTMF G729


Inband dtmf only works on alaw/ulaw. Use any other mode and it should 
work

On Fri, 22 Oct 2004 17:02:00 -0200, JOAO CARLOS MOURA
<[EMAIL PROTECTED]> wrote:
I just installed G729a my Asterisk. I am
facing some problems on DTMFMODE=INBAND. I just can t transfer my 
calls. Is
there anybody out there who could give me WHICH DTMFMODE to use?

P.S -> I already tried DTMFMODE = RFC2833 and did not work!
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Re: [Asterisk-Users] SIP termination in Brazil

[Julio Arruda]

Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and running.
Daniel
IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio 
de Janeiro.

Johannes van Hulst wrote:
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
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Re: [Asterisk-Users] E3 PCI Cards

[Julio Arruda]

Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
 

No, but if you find an E3 PCI card with nice Linux support there might 
be people interested in helping to get it working with *.
Doesn't ImageStream have these (E3 and others) cards running in Linux 
(for their routers Linux-based ?).
Still, someone mentioned horse-power AND the 'all eggs in a single E3' 
problem here...
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Re: [Asterisk-Users] Extending E1's over a Satellite link

[Julio Arruda]

Tim McKee wrote:
Guys:
 
I routinely run multiple phones over our satellite system (I'm the VP of
Network Services at SDN Global, a satellite bandwidth provider located in
Charlotte NC, US).
 
Just last week I went to West Palm Beach, FL US and turned up a 10 phone
emergency call center, complete with ACD queues for an insurance company.
We were able to run all ten phones (Cisco 7960Gs, SIP) on G.729 codecs back
to my * server in Charlotte NC US.  No special settings were required on *
or the phones.
 
The satellite system *must* support *REAL* QoS and must have jitter <
~100ms.  Traditional satellite systems have *lots* more jitter than that.
The actual latency _doesn't matter_ as long as the jitter is steady.  We are
even doing 'double-hop' phone calls successfully, where the latency is
double the normal latency.
 
Anyone that wants more detailed info contact me off-list.
Out of sheer curisity, the delay itself doesn't make the conversation 
'bad' (meaning, walkie-talkie/roger-and-over-like ?)
The codec itself should introduce some dozens of ms, but the satellite, 
is not at least 300ms one way or something like that ? adding the 
codecs, and the jitter buffers and etc..I wonder how good/bad is it ?
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[Asterisk-Users] Grandstream x Asterisk 1.0 RC1 x VOIPJet

[Julio Arruda]

Sirs/Ladies,
Not sure if anyone saw anything like that before...
I was playing with an Asterisk setup with a Grandstream BT101 (1.0.5.11) 
and www.voipjet.com (IAX2).
The other devices I have home (Sipura 3k and DTA-310) seem to work just 
fine, but the Grandstream seems to suffer from one-way voice (remote end 
can't hear me).
The only workaround I found so far (have not spoken with VOIPJet supoprt 
about this), was to configure the GS entry in sip.conf to disallow=all 
and allow=ilbc, while iax.conf has disallow=all, allow=ulaw mulaw and 
ilbc (seems the call is established in this case, with the GS leg in 
ILBC and the voipjet in ulaw).
Should I assume this would force the Asterisk to do the transcoding to 
any other end-point that doesn't handle/prefer ilbc ?
What about DTMF (I've changed to 'SIP INFO' in sip.conf and the 
grandstream web config) ?
[], 

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