Re: [asterisk-users] Asterisk w/ Nokia "e" Series Handsets

[Remco Barendse]

On Tue, 12 May 2009, Andrew Joakimsen wrote:

> Overall, given the limitations of WiFi, it works rather well. I've
> never had to reboot my E71 or play with the settings after it was
> setup. Something I can't say about other WiFi (only) phones I have
> used. And VoIP on Windows mobile phones is crap.

I installed a program called WeFi on my phone. To the phone it appears as 
one single access point, while WeFi handles connections to all access 
points automatically. It solves the problem of creating one SIP profile 
for each access point.


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Re: [asterisk-users] Rusting Snoms?

[Remco Barendse]

On Sat, 9 May 2009, Tim Panton wrote:

> This is a bit off topic, because I 'think' it isn't an Asterisk problem.
> However I'm not sure and anyhow I'm hoping someone may recognize the symptom.
>
> We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years 
> old)
> were packed up for the move, then unpacked a couple of weeks later.
>
> On unpacking them and connecting them to the new network, several of them
> didn't work well. The symptom is that outgoing RTP audio is garbled - like 
> the
> packets are pulsed. Inbound is fine. This isn't true for all of the phones,
> just some of them. (The all run the same SNOM firmware)

I didn't experience these problems, even when not using the phones for 
more than a year. The Snom 190s are pretty well built though, unlike 
the 360's that are just a piece of crap. Did you check the network 
settings, maybe there is still some NAT stuff present in the phones? I 
would do a factory reset on the phone and try again.

I bought 30 Snom 360's in 2005, since then 12 of them have been showing 
problems, an absolutely unacceptable mortality rate for a business phone.

I had 4 power supplies gone dead, of 4 phones the display went dead 
and i had about 7 phones where the receiver hook switch broke off inside 
the case (if you open the phone you see that the receiver hook switched is 
locked in between to tiny pieces of plastic). The hook switch even broke 
in my phone and i am very careful with it, i don't handle it rough...
(i'm not counting the power supplies among broken phones)

So far Snom repaired 2 batches under warranty, one after the 
warranty expired, the 3rd batch of 4 phones if waiting to be sent to 
them, will wait to see their reply, the fail rate is unacceptable and way 
beyond normal.

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Re: [asterisk-users] Beginning to use Asterisk and tests withextensions

[Dana Harding]

>> I was testing and sometimes with Echo() and MusicOnHold the sound is
>> broken. Is there some form to solve this?

> The problem with this is that although Twinkle now work, I can't have
> anything else running that uses sound because then the audio is blocked.

If the issue is/was Twinkle itself,  what about using a different softphone?
The Windows installs of Zoiper and SJphone are working fine for me with 
Asterisk.
Both offer Linux versions.

Reference your mention of testing with an SPA-3102:
I haven't worked with any SPA-3102 units -  my experience with the SPA3000 
is
outlined 
http://lists.digium.com/pipermail/asterisk-users/2006-August/162238.html
A large amount of information is available from google and voip-info.org 
regarding the configuration of these units with asterisk.


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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Paul Hales]

Alex Samad wrote:
> On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
>
>   
>> What happens if you make a call in from the old fax line and send that
>> over to the old PABX? Does that work OK?
>> 
>
> not sure what you are asking here.  I have checked an incoming call
> through the FXO(PSTN) through to a FXS port (pabx)
>
>   

Testing a phone call from the outside world, into the fax line, into the
asterisk box and then to the PABX.

This avoids all networking.

PaulH

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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Paul Hales]

Alex Samad wrote:
> On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
>   
>> Have you tried plugging analog phones into the FXS ports in the Asterisk
>> box?
>> 
>
> good ideal, but trying to find an old style phone the site has a
> commander PABX with digital handsets. I will see if I can track one down
> :)
>
> A
>
>   

You could use the fax machine (if it has a handset).
Failing that, you will have one at home.

This is most likely just a question of getting some settings right, and
an analog handset will be a quick way to check how close you are.

PaulH

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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Marco Sambo]

FXO channels shuld have FXS signalling, and FXS channels shuld have FXO
signalling, so:

# FXO channels are 1,2,3
fxsks=1,2,3
# FXS channel is 4
fxoks=4






> sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
> > that a attached fxs presents internally as a fxo
> >
> > I have a pstn line attached to the FXO and I have my pabx attached to
> > 2 FXS ports, which signal as fxo into asterisk (I could be wrong about
> > that).
>
> >>> # cat /etc/zaptel.conf
> >>> fxsks=4
> >>> fxoks=1,2,3
>
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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Alex Samad]

On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
> 
> Have you tried plugging analog phones into the FXS ports in the Asterisk
> box?

good ideal, but trying to find an old style phone the site has a
commander PABX with digital handsets. I will see if I can track one down
:)

A

> 
> That should let you know what the Asterisk is really doing with it's FXS
> ports.
> 
> PaulH
> 
> 


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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Alex Samad]

On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
> Alex Samad wrote:
> > On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
> >   
> >> I think you have your line types mixed up - FXS is for phones, FXO is
> >> for lines.
> >> 
> >
> > sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
> > that a attached fxs presents internally as a fxo
> >
> > I have a pstn line attached to the FXO and I have my pabx attached to
> > 2 FXS ports, which signal as fxo into asterisk (I could be wrong about
> > that).
> >
> >
> >   
> By reading your configs below, you could be right - ports 1,2 and 3 are
> FXS, while 4 is FXO.

fingers crossed I am.  Pretty sure I am, like I said I have been able to
make out going calls on the pstn line (although coming to think about it
I haven't actually tried talking .


> 
> What happens if you make a call in from the old fax line and send that
> over to the old PABX? Does that work OK?

not sure what you are asking here.  I have checked an incoming call
through the FXO(PSTN) through to a FXS port (pabx)

> 
> You could also buy some IP phones or put softphones around. That would
> solve the problem (you said that a softphone worked OK)

I have bought a snom to trial, but I don't want to make to many
changes in one time.

The annoying thing is the spa9000 can talk to it with its fxs ports :(

Alex

> 
> PaulH
> 
> 
> >   
> >> An analogue passthorugh setup _is_ doable, just not overly recommended.
> >>
> >> PaulH
> >>
> >>
> >> Alex Samad wrote:
> >> 
> >>> Hi
> >>>
> >>> I am in the middle of move a small business over from legacy PABX + PSTN
> >>> lines to VOIP infrastructure.
> >>>
> >>> I borrowed a spa9000 to place between the PABX and the PSTN lines. I
> >>> have had this going for a while (>5 months) and it has been working fine
> >>> (some issues with echo and other minor things), which is why I am moving
> >>> to asterisk.
> >>>
> >>> I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
> >>> and used just in case the internet connection is down.
> >>>
> >>> I have tested the pstn line connection with a soft phone and it seems to
> >>> be working fine. I need some help on how to tell asterisk to ignore the
> >>> line for incoming !
> >>>
> >>> when I connect the PABX to the FXO ports I ran into a problem.
> >>>
> >>> It seems to register okay, I pick up the handset on the pabx and select
> >>> line 1 and i can hear a dial tone (same with line2) - this is the same
> >>> what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
> >>> use.
> >>>
> >>> But I can't hear anything from the pabx - no dtmf tones and thus can't
> >>> dial!
> >>>
> >>> when I try dialing in from the internet to asterisk then to ZAP/g1 the
> >>> pabx can see the ring and I can pick up the phone I can hear the other
> >>> end, but they can't hear me.
> >>>
> >>> I don't believe its a firewall issue as I can't dial from the pabx
> >>>
> >>> okay some print outs
> >>>
> >>> # zaptel_hardware 
> >>> pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P
> >>>
> >>> # ztcfg -vv
> >>>
> >>> Zaptel Version: 1.4.11
> >>> Echo Canceller: MG2
> >>> Configuration
> >>> ==
> >>>
> >>>
> >>> Channel map:
> >>>
> >>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> >>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> >>> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> >>> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> >>>
> >>> 4 channels to configure.
> >>>
> >>> # cat /etc/zaptel.conf 
> >>> fxsks=4
> >>> fxoks=1,2,3
> >>>
> >>> loadzone=au
> >>> defaultzone=au
> >>>
> >>> /etc/asterisk/zapata.conf
> >>> 
> >>> # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
> >>> [trunkgroups]
> >>> [channels]
> >>> context=default
> >>> switchtype=national
> >>> signalling=fxo_ks
> >>> rxwink=300; Atlas seems to use long (250ms) winks
> >>> usecallerid=yes
> >>> hidecallerid=no
> >>> callwaiting=yes
> >>> usecallingpres=yes
> >>> callwaitingcallerid=yes
> >>> threewaycalling=yes
> >>> transfer=yes
> >>> canpark=yes
> >>> cancallforward=yes
> >>> callreturn=yes
> >>> echocancel=yes
> >>> echocancelwhenbridged=yes
> >>> rxgain=0.0
> >>> txgain=0.0
> >>> group=1
> >>> callgroup=1
> >>> pickupgroup=1
> >>> immediate=no
> >>> usecallerid=yes
> >>> hidecallerid=no
> >>> callwaiting=yes
> >>> threewaycalling=yes
> >>> transfer=yes
> >>> echocancel=yes
> >>> echocancelwhenbridged=yes
> >>> rxgain=0.0
> >>> txgain=0.0
> >>> Group=1
> >>> signalling=fxo_ks
> >>> context=in-pbx
> >>> channel=1-2
> >>> Group=2
> >>> echocancel=yes
> >>> signalling=fxs_ks
> >>> context=in-pstn
> >>> channel=4
> >>> Group=3
> >>> signalling=fxo_ks
> >>> context=in-spare
> >>> channel=3
> >>>
> >>>
> >>> the thing that has me beet is that it work with the spa9000 I would
> >>> expect it to just sort of work with the digium card.
> >>>
> >>> the os is debian amd64 2.6.26
> >>> #dpkg -l asteri*

Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Paul Hales]

Have you tried plugging analog phones into the FXS ports in the Asterisk
box?

That should let you know what the Asterisk is really doing with it's FXS
ports.

PaulH


Alex Samad wrote:
> On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
>   
>> I think you have your line types mixed up - FXS is for phones, FXO is
>> for lines.
>> 
>
> sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
> that a attached fxs presents internally as a fxo
>
> I have a pstn line attached to the FXO and I have my pabx attached to
> 2 FXS ports, which signal as fxo into asterisk (I could be wrong about
> that).
>
>
>
>   
>> An analogue passthorugh setup _is_ doable, just not overly recommended.
>>
>> PaulH
>>
>>
>> Alex Samad wrote:
>> 
>>> Hi
>>>
>>> I am in the middle of move a small business over from legacy PABX + PSTN
>>> lines to VOIP infrastructure.
>>>
>>> I borrowed a spa9000 to place between the PABX and the PSTN lines. I
>>> have had this going for a while (>5 months) and it has been working fine
>>> (some issues with echo and other minor things), which is why I am moving
>>> to asterisk.
>>>
>>> I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
>>> and used just in case the internet connection is down.
>>>
>>> I have tested the pstn line connection with a soft phone and it seems to
>>> be working fine. I need some help on how to tell asterisk to ignore the
>>> line for incoming !
>>>
>>> when I connect the PABX to the FXO ports I ran into a problem.
>>>
>>> It seems to register okay, I pick up the handset on the pabx and select
>>> line 1 and i can hear a dial tone (same with line2) - this is the same
>>> what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
>>> use.
>>>
>>> But I can't hear anything from the pabx - no dtmf tones and thus can't
>>> dial!
>>>
>>> when I try dialing in from the internet to asterisk then to ZAP/g1 the
>>> pabx can see the ring and I can pick up the phone I can hear the other
>>> end, but they can't hear me.
>>>
>>> I don't believe its a firewall issue as I can't dial from the pabx
>>>
>>> okay some print outs
>>>
>>> # zaptel_hardware 
>>> pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P
>>>
>>> # ztcfg -vv
>>>
>>> Zaptel Version: 1.4.11
>>> Echo Canceller: MG2
>>> Configuration
>>> ==
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
>>> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>>>
>>> 4 channels to configure.
>>>
>>> # cat /etc/zaptel.conf 
>>> fxsks=4
>>> fxoks=1,2,3
>>>
>>> loadzone=au
>>> defaultzone=au
>>>
>>> /etc/asterisk/zapata.conf
>>> 
>>> # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
>>> [trunkgroups]
>>> [channels]
>>> context=default
>>> switchtype=national
>>> signalling=fxo_ks
>>> rxwink=300  ; Atlas seems to use long (250ms) winks
>>> usecallerid=yes
>>> hidecallerid=no
>>> callwaiting=yes
>>> usecallingpres=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> canpark=yes
>>> cancallforward=yes
>>> callreturn=yes
>>> echocancel=yes
>>> echocancelwhenbridged=yes
>>> rxgain=0.0
>>> txgain=0.0
>>> group=1
>>> callgroup=1
>>> pickupgroup=1
>>> immediate=no
>>> usecallerid=yes
>>> hidecallerid=no
>>> callwaiting=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> echocancel=yes
>>> echocancelwhenbridged=yes
>>> rxgain=0.0
>>> txgain=0.0
>>> Group=1
>>> signalling=fxo_ks
>>> context=in-pbx
>>> channel=1-2
>>> Group=2
>>> echocancel=yes
>>> signalling=fxs_ks
>>> context=in-pstn
>>> channel=4
>>> Group=3
>>> signalling=fxo_ks
>>> context=in-spare
>>> channel=3
>>>
>>>
>>> the thing that has me beet is that it work with the spa9000 I would
>>> expect it to just sort of work with the digium card.
>>>
>>> the os is debian amd64 2.6.26
>>> #dpkg -l asteri* | grep ^ii
>>> ii  asterisk1:1.4.21.2~dfsg-3
>>> Open Source Private Branch Exchange (PBX)
>>> ii  asterisk-barbarast.com  0.0.0-1
>>> asterisk setup for hme1.samad.com.au
>>> ii  asterisk-doc1:1.4.21.2~dfsg-3
>>> Source code documentation for Asterisk
>>> ii  asterisk-sounds-extra   1.4.7-1
>>> Additional sound files for the Asterisk PBX
>>> ii  asterisk-sounds-main1:1.4.21.2~dfsg-3
>>> Core Sound files for Asterisk (English)
>>>
>>> #dpkg -l zapt* | grep ^ii
>>> ii  zaptel  1:1.4.11~dfsg-3
>>> zapata telephony utilities
>>> ii  zaptel-modules-2.6.22-2-amd64   1:1.4.11~dfsg-3+2.6.22-4
>>> zaptel modules for Linux (kernel 2.6.22-2-am
>>> ii  zaptel-modules-2.6.26-2-amd64
>>> 1:1.4.11~dfsg-3+2.6.26-15   zaptel modules for Linux (kernel 2.6.26-2-am
>>> ii  zaptel-source
>>>
>>>
>>> thanks
>>> Alex
>>>
>>>   
>>> --

Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Paul Hales]

Alex Samad wrote:
> On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
>   
>> I think you have your line types mixed up - FXS is for phones, FXO is
>> for lines.
>> 
>
> sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
> that a attached fxs presents internally as a fxo
>
> I have a pstn line attached to the FXO and I have my pabx attached to
> 2 FXS ports, which signal as fxo into asterisk (I could be wrong about
> that).
>
>
>   
By reading your configs below, you could be right - ports 1,2 and 3 are
FXS, while 4 is FXO.

What happens if you make a call in from the old fax line and send that
over to the old PABX? Does that work OK?

You could also buy some IP phones or put softphones around. That would
solve the problem (you said that a softphone worked OK)

PaulH


>   
>> An analogue passthorugh setup _is_ doable, just not overly recommended.
>>
>> PaulH
>>
>>
>> Alex Samad wrote:
>> 
>>> Hi
>>>
>>> I am in the middle of move a small business over from legacy PABX + PSTN
>>> lines to VOIP infrastructure.
>>>
>>> I borrowed a spa9000 to place between the PABX and the PSTN lines. I
>>> have had this going for a while (>5 months) and it has been working fine
>>> (some issues with echo and other minor things), which is why I am moving
>>> to asterisk.
>>>
>>> I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
>>> and used just in case the internet connection is down.
>>>
>>> I have tested the pstn line connection with a soft phone and it seems to
>>> be working fine. I need some help on how to tell asterisk to ignore the
>>> line for incoming !
>>>
>>> when I connect the PABX to the FXO ports I ran into a problem.
>>>
>>> It seems to register okay, I pick up the handset on the pabx and select
>>> line 1 and i can hear a dial tone (same with line2) - this is the same
>>> what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
>>> use.
>>>
>>> But I can't hear anything from the pabx - no dtmf tones and thus can't
>>> dial!
>>>
>>> when I try dialing in from the internet to asterisk then to ZAP/g1 the
>>> pabx can see the ring and I can pick up the phone I can hear the other
>>> end, but they can't hear me.
>>>
>>> I don't believe its a firewall issue as I can't dial from the pabx
>>>
>>> okay some print outs
>>>
>>> # zaptel_hardware 
>>> pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P
>>>
>>> # ztcfg -vv
>>>
>>> Zaptel Version: 1.4.11
>>> Echo Canceller: MG2
>>> Configuration
>>> ==
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
>>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
>>> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>>>
>>> 4 channels to configure.
>>>
>>> # cat /etc/zaptel.conf 
>>> fxsks=4
>>> fxoks=1,2,3
>>>
>>> loadzone=au
>>> defaultzone=au
>>>
>>> /etc/asterisk/zapata.conf
>>> 
>>> # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
>>> [trunkgroups]
>>> [channels]
>>> context=default
>>> switchtype=national
>>> signalling=fxo_ks
>>> rxwink=300  ; Atlas seems to use long (250ms) winks
>>> usecallerid=yes
>>> hidecallerid=no
>>> callwaiting=yes
>>> usecallingpres=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> canpark=yes
>>> cancallforward=yes
>>> callreturn=yes
>>> echocancel=yes
>>> echocancelwhenbridged=yes
>>> rxgain=0.0
>>> txgain=0.0
>>> group=1
>>> callgroup=1
>>> pickupgroup=1
>>> immediate=no
>>> usecallerid=yes
>>> hidecallerid=no
>>> callwaiting=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> echocancel=yes
>>> echocancelwhenbridged=yes
>>> rxgain=0.0
>>> txgain=0.0
>>> Group=1
>>> signalling=fxo_ks
>>> context=in-pbx
>>> channel=1-2
>>> Group=2
>>> echocancel=yes
>>> signalling=fxs_ks
>>> context=in-pstn
>>> channel=4
>>> Group=3
>>> signalling=fxo_ks
>>> context=in-spare
>>> channel=3
>>>
>>>
>>> the thing that has me beet is that it work with the spa9000 I would
>>> expect it to just sort of work with the digium card.
>>>
>>> the os is debian amd64 2.6.26
>>> #dpkg -l asteri* | grep ^ii
>>> ii  asterisk1:1.4.21.2~dfsg-3
>>> Open Source Private Branch Exchange (PBX)
>>> ii  asterisk-barbarast.com  0.0.0-1
>>> asterisk setup for hme1.samad.com.au
>>> ii  asterisk-doc1:1.4.21.2~dfsg-3
>>> Source code documentation for Asterisk
>>> ii  asterisk-sounds-extra   1.4.7-1
>>> Additional sound files for the Asterisk PBX
>>> ii  asterisk-sounds-main1:1.4.21.2~dfsg-3
>>> Core Sound files for Asterisk (English)
>>>
>>> #dpkg -l zapt* | grep ^ii
>>> ii  zaptel  1:1.4.11~dfsg-3
>>> zapata telephony utilities
>>> ii  zaptel-modules-2.6.22-2-amd64   1:1.4.11~dfsg-3+2.6.22-4
>>> zaptel modules for Linux (kernel 2.6.22-2-am
>>> ii  zaptel-modules-2.6.26

Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Alex Samad]

On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
> 
> I think you have your line types mixed up - FXS is for phones, FXO is
> for lines.

sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo

I have a pstn line attached to the FXO and I have my pabx attached to
2 FXS ports, which signal as fxo into asterisk (I could be wrong about
that).



> 
> An analogue passthorugh setup _is_ doable, just not overly recommended.
> 
> PaulH
> 
> 
> Alex Samad wrote:
> > Hi
> >
> > I am in the middle of move a small business over from legacy PABX + PSTN
> > lines to VOIP infrastructure.
> >
> > I borrowed a spa9000 to place between the PABX and the PSTN lines. I
> > have had this going for a while (>5 months) and it has been working fine
> > (some issues with echo and other minor things), which is why I am moving
> > to asterisk.
> >
> > I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
> > and used just in case the internet connection is down.
> >
> > I have tested the pstn line connection with a soft phone and it seems to
> > be working fine. I need some help on how to tell asterisk to ignore the
> > line for incoming !
> >
> > when I connect the PABX to the FXO ports I ran into a problem.
> >
> > It seems to register okay, I pick up the handset on the pabx and select
> > line 1 and i can hear a dial tone (same with line2) - this is the same
> > what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
> > use.
> >
> > But I can't hear anything from the pabx - no dtmf tones and thus can't
> > dial!
> >
> > when I try dialing in from the internet to asterisk then to ZAP/g1 the
> > pabx can see the ring and I can pick up the phone I can hear the other
> > end, but they can't hear me.
> >
> > I don't believe its a firewall issue as I can't dial from the pabx
> >
> > okay some print outs
> >
> > # zaptel_hardware 
> > pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P
> >
> > # ztcfg -vv
> >
> > Zaptel Version: 1.4.11
> > Echo Canceller: MG2
> > Configuration
> > ==
> >
> >
> > Channel map:
> >
> > Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> > Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> > Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> > Channel 04: FXS Kewlstart (Default) (Slaves: 04)
> >
> > 4 channels to configure.
> >
> > # cat /etc/zaptel.conf 
> > fxsks=4
> > fxoks=1,2,3
> >
> > loadzone=au
> > defaultzone=au
> >
> > /etc/asterisk/zapata.conf
> > 
> > # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
> > [trunkgroups]
> > [channels]
> > context=default
> > switchtype=national
> > signalling=fxo_ks
> > rxwink=300  ; Atlas seems to use long (250ms) winks
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > usecallingpres=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > canpark=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > group=1
> > callgroup=1
> > pickupgroup=1
> > immediate=no
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > threewaycalling=yes
> > transfer=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > Group=1
> > signalling=fxo_ks
> > context=in-pbx
> > channel=1-2
> > Group=2
> > echocancel=yes
> > signalling=fxs_ks
> > context=in-pstn
> > channel=4
> > Group=3
> > signalling=fxo_ks
> > context=in-spare
> > channel=3
> >
> >
> > the thing that has me beet is that it work with the spa9000 I would
> > expect it to just sort of work with the digium card.
> >
> > the os is debian amd64 2.6.26
> > #dpkg -l asteri* | grep ^ii
> > ii  asterisk1:1.4.21.2~dfsg-3
> > Open Source Private Branch Exchange (PBX)
> > ii  asterisk-barbarast.com  0.0.0-1
> > asterisk setup for hme1.samad.com.au
> > ii  asterisk-doc1:1.4.21.2~dfsg-3
> > Source code documentation for Asterisk
> > ii  asterisk-sounds-extra   1.4.7-1
> > Additional sound files for the Asterisk PBX
> > ii  asterisk-sounds-main1:1.4.21.2~dfsg-3
> > Core Sound files for Asterisk (English)
> >
> > #dpkg -l zapt* | grep ^ii
> > ii  zaptel  1:1.4.11~dfsg-3
> > zapata telephony utilities
> > ii  zaptel-modules-2.6.22-2-amd64   1:1.4.11~dfsg-3+2.6.22-4
> > zaptel modules for Linux (kernel 2.6.22-2-am
> > ii  zaptel-modules-2.6.26-2-amd64
> > 1:1.4.11~dfsg-3+2.6.26-15   zaptel modules for Linux (kernel 2.6.26-2-am
> > ii  zaptel-source
> >
> >
> > thanks
> > Alex
> >
> >   
> > 
> >
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[asterisk-users] dtmf issues?

[Jerome Deyle]

Ok, I'm still rather new to Asterisk, and I'm sure there is a simple fix 
here, but I can't see it.

Client with a small system, AsteriskNow 1.4, 10 Polycom IP330 phones. Has 
been up and running flawlessly for about a year.
This morning I logged in to make a couple of extension changes, and noticed 
in FreePBX that there were a few module updates available. Downloaded and 
installed, with no errors.

Now users are reporting that ComedianMail is not recognizing the "3" digit, 
for example, the option to change/record your greetings. All other tones 
seem to be recognized correctly. Phones are all configured dfmtmode=rfc2833, 
as are the extensions as configured in FreePBX.

Can someone steer me in the right direction??

Related question.In CentOS, where are package updates logged??  I'd like 
to review what was updated this morning.

Thanks in advance.
Jerome Deyle
 


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Re: [asterisk-users] Help need to do Lookup from odbc database

[Tilghman Lesher]

On Wednesday 13 May 2009 17:55:41 carl Lougher wrote:
> Howdy,
> How do i perform a lookup from a remote odbc database in the asterisk
> dialplan?
>
> I can do it with mysql but not sure of commands for odbc connection.

See func_odbc.conf for examples.  You'll also need to setup res_odbc.conf, as
this is where func_odbc obtains its handles.

-- 
Tilghman

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Re: [asterisk-users] Add Monitor application to call suppresses audio

[Barry L. Kline]

David Backeberg wrote:

> I don't know why recording is breaking your calls. My guess is
> something is screwed up with your PRI configuration. Are you getting
> alarms in your logs from dahdi?

Not a peep, either with or without using the monitor command.   I've
been using this system for around four months during which time it has
performed flawlessly, running through 20K calls.

> You should try to reproduce the problem on demand by generalizing your
> dialplan, change the number of the answer service to the number of
> your cell phone, and run some calls through.

Done.

> I've been recording calls with 1.6.0 series using MixMonitor() and
> haven't been having problems, making me think the recordings step is
> coincidental. Crank up the verbosity, run some calls through and tell
> us what's happening.

To avoid wrapping, I've posted the results from my tests to this link:
http://www.pastebin.ca/1422291

The first call is the dialplan I have been using which works perfectly.

The second call, which worked, was the first attempt to use Monitor(),
after having restarted Asterisk.

The third call is another attempt at getting a recording.   It, and any
subsequent call, fails miserably.

I'm open for any suggestions.  Thanks very much David.

Barry


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Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

[Paul Hales]

I think you have your line types mixed up - FXS is for phones, FXO is
for lines.

An analogue passthorugh setup _is_ doable, just not overly recommended.

PaulH


Alex Samad wrote:
> Hi
>
> I am in the middle of move a small business over from legacy PABX + PSTN
> lines to VOIP infrastructure.
>
> I borrowed a spa9000 to place between the PABX and the PSTN lines. I
> have had this going for a while (>5 months) and it has been working fine
> (some issues with echo and other minor things), which is why I am moving
> to asterisk.
>
> I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
> and used just in case the internet connection is down.
>
> I have tested the pstn line connection with a soft phone and it seems to
> be working fine. I need some help on how to tell asterisk to ignore the
> line for incoming !
>
> when I connect the PABX to the FXO ports I ran into a problem.
>
> It seems to register okay, I pick up the handset on the pabx and select
> line 1 and i can hear a dial tone (same with line2) - this is the same
> what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
> use.
>
> But I can't hear anything from the pabx - no dtmf tones and thus can't
> dial!
>
> when I try dialing in from the internet to asterisk then to ZAP/g1 the
> pabx can see the ring and I can pick up the phone I can hear the other
> end, but they can't hear me.
>
> I don't believe its a firewall issue as I can't dial from the pabx
>
> okay some print outs
>
> # zaptel_hardware 
> pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P
>
> # ztcfg -vv
>
> Zaptel Version: 1.4.11
> Echo Canceller: MG2
> Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)
>
> 4 channels to configure.
>
> # cat /etc/zaptel.conf 
> fxsks=4
> fxoks=1,2,3
>
> loadzone=au
> defaultzone=au
>
> /etc/asterisk/zapata.conf
> 
> # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
> [trunkgroups]
> [channels]
> context=default
> switchtype=national
> signalling=fxo_ks
> rxwink=300; Atlas seems to use long (250ms) winks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> Group=1
> signalling=fxo_ks
> context=in-pbx
> channel=1-2
> Group=2
> echocancel=yes
> signalling=fxs_ks
> context=in-pstn
> channel=4
> Group=3
> signalling=fxo_ks
> context=in-spare
> channel=3
>
>
> the thing that has me beet is that it work with the spa9000 I would
> expect it to just sort of work with the digium card.
>
> the os is debian amd64 2.6.26
> #dpkg -l asteri* | grep ^ii
> ii  asterisk1:1.4.21.2~dfsg-3
> Open Source Private Branch Exchange (PBX)
> ii  asterisk-barbarast.com  0.0.0-1
> asterisk setup for hme1.samad.com.au
> ii  asterisk-doc1:1.4.21.2~dfsg-3
> Source code documentation for Asterisk
> ii  asterisk-sounds-extra   1.4.7-1
> Additional sound files for the Asterisk PBX
> ii  asterisk-sounds-main1:1.4.21.2~dfsg-3
> Core Sound files for Asterisk (English)
>
> #dpkg -l zapt* | grep ^ii
> ii  zaptel  1:1.4.11~dfsg-3
> zapata telephony utilities
> ii  zaptel-modules-2.6.22-2-amd64   1:1.4.11~dfsg-3+2.6.22-4
> zaptel modules for Linux (kernel 2.6.22-2-am
> ii  zaptel-modules-2.6.26-2-amd64
> 1:1.4.11~dfsg-3+2.6.26-15   zaptel modules for Linux (kernel 2.6.26-2-am
> ii  zaptel-source
>
>
> thanks
> Alex
>
>   
> 
>
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Re: [asterisk-users] Add Monitor application to call suppresses audio

[David Backeberg]

On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline  wrote:
> If I insert a Monitor() prior to dialing the outbound call, I get no
> audio in the recording and the caller hears no audio.   Occasionally it
> works (perhaps 1 out of 5 times) but most of the time the caller can't
> hear the callee, and vice versa.
> I'm using Asterisk 1.6.0.9,  LIBPRI 1.4.10,  and DAHDI 2.1.0.4.
> Can anyone shine any light on why this problem is occurring?

I don't know why recording is breaking your calls. My guess is
something is screwed up with your PRI configuration. Are you getting
alarms in your logs from dahdi?

You should try to reproduce the problem on demand by generalizing your
dialplan, change the number of the answer service to the number of
your cell phone, and run some calls through.

I've been recording calls with 1.6.0 series using MixMonitor() and
haven't been having problems, making me think the recordings step is
coincidental. Crank up the verbosity, run some calls through and tell
us what's happening.

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[asterisk-users] Problem with Asterisk + TDM410 FXO

[Alex Samad]

Hi

I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.

I borrowed a spa9000 to place between the PABX and the PSTN lines. I
have had this going for a while (>5 months) and it has been working fine
(some issues with echo and other minor things), which is why I am moving
to asterisk.

I bought a tdm410 with 3 fxo + fxs.  The fxs is connected to a fax line
and used just in case the internet connection is down.

I have tested the pstn line connection with a soft phone and it seems to
be working fine. I need some help on how to tell asterisk to ignore the
line for incoming !

when I connect the PABX to the FXO ports I ran into a problem.

It seems to register okay, I pick up the handset on the pabx and select
line 1 and i can hear a dial tone (same with line2) - this is the same
what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in
use.

But I can't hear anything from the pabx - no dtmf tones and thus can't
dial!

when I try dialing in from the internet to asterisk then to ZAP/g1 the
pabx can see the ring and I can pick up the phone I can hear the other
end, but they can't hear me.

I don't believe its a firewall issue as I can't dial from the pabx

okay some print outs

# zaptel_hardware 
pci::05:02.0 wctdm24xxp+  d161:8005 Wildcard TDM410P

# ztcfg -vv

Zaptel Version: 1.4.11
Echo Canceller: MG2
Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels to configure.

# cat /etc/zaptel.conf 
fxsks=4
fxoks=1,2,3

loadzone=au
defaultzone=au

/etc/asterisk/zapata.conf

# grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$'
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
Group=1
signalling=fxo_ks
context=in-pbx
channel=1-2
Group=2
echocancel=yes
signalling=fxs_ks
context=in-pstn
channel=4
Group=3
signalling=fxo_ks
context=in-spare
channel=3


the thing that has me beet is that it work with the spa9000 I would
expect it to just sort of work with the digium card.

the os is debian amd64 2.6.26
#dpkg -l asteri* | grep ^ii
ii  asterisk1:1.4.21.2~dfsg-3
Open Source Private Branch Exchange (PBX)
ii  asterisk-barbarast.com  0.0.0-1
asterisk setup for hme1.samad.com.au
ii  asterisk-doc1:1.4.21.2~dfsg-3
Source code documentation for Asterisk
ii  asterisk-sounds-extra   1.4.7-1
Additional sound files for the Asterisk PBX
ii  asterisk-sounds-main1:1.4.21.2~dfsg-3
Core Sound files for Asterisk (English)

#dpkg -l zapt* | grep ^ii
ii  zaptel  1:1.4.11~dfsg-3
zapata telephony utilities
ii  zaptel-modules-2.6.22-2-amd64   1:1.4.11~dfsg-3+2.6.22-4
zaptel modules for Linux (kernel 2.6.22-2-am
ii  zaptel-modules-2.6.26-2-amd64
1:1.4.11~dfsg-3+2.6.26-15   zaptel modules for Linux (kernel 2.6.26-2-am
ii  zaptel-source


thanks
Alex



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[asterisk-users] Help need to do Lookup from odbc database

[carl Lougher]

Howdy,
How do i perform a lookup from a remote odbc database in the asterisk dialplan?

I can do it with mysql but not sure of commands for odbc connection.

Cheers!!!


  

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Re: [asterisk-users] [asterisk-user] Which policy for ISDN BRI support in NT/PtMP ?

[Kristijan Vrban]

good news, i just made my isdn device ring! ok, after it ring, any
timout then hangup up the chan, but a ringing from chan_dahdi via
bri_net_ptmp -> isdn_device was possible.

to made this happen i made some very crude hacks inside libpri, but i
hope the next days i can offer a patch that offer some basic nt_ptmp
functionality. Stay tuned :)

Kristijan

2009/5/12, Tzafrir Cohen :
> On Tue, May 12, 2009 at 08:05:49AM +0200, Olivier wrote:
>> 2009/5/12 Kristijan Vrban 
>>
>> > For those also need NT over PtMP, i started a initial patch for it. Very
>> > limited at the moment, only one incoming call to chan_dahdi from one
>> > device is possible. But i was pleasantly surprised that NT-ptmp is
>> > working
>> > anyway
>> >
>> > Get the patch here: http://bugs.digium.com/view.php?id=15048
>
> Or rather: works in one direction: calls from a phone to the NT work.
> Calls in the other way don't make it.
>
> I believe it's much better than nothing, though, and I'm testing this
> patch in the new debs I have.
>
>> That is great news !!!
>>
>> How best can we contribute to make this happen ?
>
> Test this. Report how it (mis)behaves. And maybe try to trace why calls
> from the NT side don't get through.
>
>> Will the output most probably be a new libpri 1.4.X (or 1.6.X) or will it
>> also include a new Asterisk version ?
>
> For starters there will likely be some changes required in libpri .
> There is no libpri 1.6.x and not likely to be one in the near future.
> The "trunk" of libpri is branches/1.4 .
>
> --
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Double dial.

[Steve Edwards]

Date: Thu, 14 May 2009 01:59:58 +0300
From: Catalin S. 
Reply-To: Asterisk Developers Mailing List 
To: Asterisk Developers Mailing List 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-dev] Double dial.

Date: Thu, 14 May 2009 01:59:58 +0300
From: Catalin S. 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

To: Asterisk Developers Mailing List 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Double dial.

Double posting?

On Thu, 14 May 2009, Catalin S. wrote:

> I have a strange situation with an SPA3102 FXO/FXS device. I'm in
> situation that when i receive a call from PBX line I must forward the
> calls to 2 VoIP numbers.
> Right now i have the following settings: (S0<:1...@gw1>). I want to
> forward at 1020 too. I tested (S0<:1010|1...@gw1>)  and doesn't work.
> Did you have any other ideea?

Please try a SPA3102 mailing list. This doesn't appear to have anything to 
do with Asterisk either on a user or dev level.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Double dial.

[Catalin S.]

Hello,

I have a strange situation with an SPA3102 FXO/FXS device. I'm in
situation that when i receive a call from PBX line I must forward the
calls to 2 VoIP numbers.
Right now i have the following settings: (S0<:1...@gw1>). I want to
forward at 1020 too. I tested (S0<:1010|1...@gw1>)  and doesn't work.
Did you have any other ideea?

Thank you.

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[asterisk-users] High Volume US Traffic? Claim DIP Compensation!

[Marco [voicetermination.org]]

This could be a nice opportunity for users with a high volume of SIP traffic
terminating in the US: 
Collecting dip fees on outbound phone calls - fees that would otherwise go
to the local phone company.




With all the recent fees and surcharges, the cost of wholesale telecom and
dialer traffic keeps rising. But what many companies with a high volume of
IP based voice traffic don't realize is that they are able to share in these
dip fees. There is no need to switch routes or carriers in order to
participate.

 

There is a minimum of roughly 300k calls per month that terminate in the US
in order to participate. I would like to ask if you would be interested in
talking about this, if so what would be a good time and number to reach you?


 

p.s. this is not some kind of fishy scheme but a way to benefit and collect
from government telecom regulations that exist.

 

Warm Regards,

 

Marco Wind

dipfees.com

 

Ph: 646-736-7816

Tf:  888-780-0253

F :  (347) 626-2242

 

 

 

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Re: [asterisk-users] High Volume US Traffic? Claim DIP Compensation!

[ContactTel Business]

You should send that to the Business list, not
users.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco
[voicetermination.org]
Sent: May-13-09 5:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] High Volume US Traffic? Claim DIP Compensation!

 

This could be a nice opportunity for users with a high volume of SIP traffic
terminating in the US: 
Collecting dip fees on outbound phone calls - fees that would otherwise go
to the local phone company.



With all the recent fees and surcharges, the cost of wholesale telecom and
dialer traffic keeps rising. But what many companies with a high volume of
IP based voice traffic don't realize is that they are able to share in these
dip fees. There is no need to switch routes or carriers in order to
participate.

 

There is a minimum of roughly 300k calls per month that terminate in the US
in order to participate. I would like to ask if you would be interested in
talking about this, if so what would be a good time and number to reach you?


 

p.s. this is not some kind of fishy scheme but a way to benefit and collect
from government telecom regulations that exist.

 

Warm Regards,

 

Marco Wind

dipfees.com

 

Ph: 646-736-7816

Tf:  888-780-0253

F :  (347) 626-2242

 

 

 

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Re: [asterisk-users] #-all.gsm

[Steve Edwards]

On Thu, 14 May 2009, David @ULC wrote:

> Recently I noted few recording link with # sign on it.

Asterisk creates links?

> like 223345#-all.gsm
>
> and all those voice files are NOT available for download.

Asterisk downloads files?

> I tried changing the file name in the mysql db and removed # but still its
> not available.

Asterisk is part of MySQL?

I thought this was an Asterisk mailing list. Oh wait -- it is :)

> What could be the reason for # and why its NOT available for download ?

Maybe the "magic number" above is something a caller entered and 
terminated input with # and your dialplan didn't remove it. Maybe your web 
server is broken and doesn't like #. Maybe some script that munges your 
files considers # to be the start of a comment.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] #-all.gsm

[David @ULC]

Recently I noted few recording link with # sign on it.

like 223345#-all.gsm

and all those voice files are NOT available for download.

I tried changing the file name in the mysql db and removed # but still its
not available.

What could be the reason for # and why its NOT available for download ?
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Re: [asterisk-users] Proxying from one server to another

[Gavin Henry]

Why not use OpenSIPS or Kamailio in stateful mode?

You will need to look at how media is handled though, but a SIP proxy
will work easily.

On 13/05/2009, Adrian Marsh  wrote:
> Hi David,
>
>
>
> Thanks for the reply. That's pretty much what I've already tried, but
> with no luck on the production machines.  In testing it worked, but the
> public IPs and single NICs were causing issues (we believe)
>
> So I was looking for a proxy-type solution.
>
>
>
> Adrian
>
>
>
> 
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Gibbons
> Sent: 13 May 2009 15:37
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Proxying from one server to another
>
>
>
> Redirect traffic with iptables like this:
>
>
>
> Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
> NEW_PUBLIC_IP
>
>
>
> I'm not sure if this will work for SIP. You may need the proxy to change
> info in the sip messages between server and client.
>
>
>
> --Dave
>
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
> Marsh
> Sent: Wednesday, May 13, 2009 8:55 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Proxying from one server to another
>
>
>
> Hi All,
>
>
>
> I'm trying to find a software package to do the following sip proxy
> work:
>
>
>
> I've an A*k server A that needs to be decommissioned, from the USA, and
> replaced by server B, in the UK. Both servers are on public internet
> IPs.
>
> Whilst the client migration happens, I want to divert all the Register
> traffic from Server A to Server B to catch any clients still left out
> there.
>
>
>
> Unfortunately, the original Clients were configured with static IPs
> instead of DNS names for the SIP Registrar, so I have to proxy Server A
> until all the clients have been updated (which might be a long time).
>
>
>
> Obviously A*k itself wont do this (as far as I know).  I've looked at
> siproxyd and party-sip, but with no success so far.
>
> I've also tried using IPtables to redirect at the IP level, but the
> public IP ranges seem to stop me from achieving this. It works in my
> local-lan testing, but not on the public servers.
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
>
>
> Adrian
>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/

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Re: [asterisk-users] Asterisk+a2billing for over 10,000 ext

[Tim Nelson]

- "James Mutuku"  wrote: 
> Hellos, 

> 
I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale. 
Has anyone done this before. What are the hardware requirements and challenges? 

> 
James 

If you're asking that sort of question, you probably shouldn't be doing it. The 
Internet already has enough 'VoIP/Telephony providers' that don't have a clue 
to what they're doing. 

--Tim 
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[asterisk-users] Asterisk+a2billing for over 10,000 ext

[James Mutuku]

Hellos,
I want to setup Asterisk+a2billing for over 10,000 extensions for voip
resale. Has anyone done this before. What are the hardware requirements and
challenges?

James
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Re: [asterisk-users] Voicemail and remote directory with SSHFS

[David Gibbons]

Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once 
of those more ubiquitous standards.

-Dave

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, May 13, 2009 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail and remote directory with SSHFS

Hello!

I am trying to mount a remote directory for voicemail using sshfs.  However, 
whenever Asterisk attempts to write the file, it fails, because SSHFS cannot 
lock the directory.  Is there a solution to this problem or an alternative 
method for using a remote directory for voicemail?

Thanks,
Elliot
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[asterisk-users] Why asterisk changes RTP destination port when it receives first RTP packet in opposite direction despite canreinvite=no

[rob.r374]

Hi,

I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP 
trunk. Since recognition didn't work correctly, I've troubleshot with 
Wireshark and saw that RTP stream is first send to one port on SIP trunk and 
then when first RTP packet arrives in opposite direction (from TTS part of 
Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port - 
therefore changing destination port.

I'm calling sip trunk with IAX2 client, have put canreinvite=no in sip.conf 
and in trunk definition

What am I doing wrong ?
Anyone else tried Zanzibar (for those not familiar - it's a first open 
source based way of doing speech recognition and synthesis in convenient 
way) ?


Thanks in advance,

regards,

Rob.

Trunk definition :
[Zanzibar]

type=peer

host=192.168.0.50

port=5090

dtmfmode=info

canreinvite=no

qualify=no

Extension definition for calling IVR :

exten => 510,1,Answer

exten => 510,2,SIPAddHeader(x-channel:${CHANNEL})

exten => 510,3,SIPAddHeader(x-application:beanId|Parrot)

exten => 510,4,Dial(SIP/Zanzibar)






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Re: [asterisk-users] Voicemail and remote directory with SSHFS

[Danny Nicholas]

Probably a permissions problem.  Check out this article

 

http://ubuntu.wordpress.com/2005/10/28/how-to-mount-a-remote-ssh-filesystem-
using-sshfs/

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, May 13, 2009 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail and remote directory with SSHFS

 

Hello!

 

I am trying to mount a remote directory for voicemail using sshfs.  However,
whenever Asterisk attempts to write the file, it fails, because SSHFS cannot
lock the directory.  Is there a solution to this problem or an alternative
method for using a remote directory for voicemail?

 

Thanks,

Elliot

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[asterisk-users] Voicemail and remote directory with SSHFS

[Elliot Murdock]

Hello!

I am trying to mount a remote directory for voicemail using sshfs.  However,
whenever Asterisk attempts to write the file, it fails, because SSHFS cannot
lock the directory.  Is there a solution to this problem or an alternative
method for using a remote directory for voicemail?

Thanks,
Elliot
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Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice router

[David Backeberg]

On Wed, May 13, 2009 at 9:21 AM, Jon Schøpzinsky  wrote:
> I used wireshark to debug the problem, and I can see that the cisco equipment 
> is correctly sending t.38 packets to asterisk, and the whole re-invite 
> process is successful.
> The problem is, that Asterisk discards the t.38 packets with the error 
> message I sent, and therefore the T.38 session never gets underway. Asterisk 
> is stuck on the same SEQ id, as it never receives anything from the cisco.
> Ive also checked that this isn't a network issue. The packets are coming 
> through, asterisk just throws them away with the error message I described.

Does any SIP traffic of any kind work properly between the machines?
How about a simple voice phone call?

Sending an audio fax over SIP would be a good test.
Posting the Cisco dialpeer config would also be helpful.

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[asterisk-users] AstriCon 2009 speaker submissions open!

[John Todd]

  AstriCon 2009 is still 5 months away, but the time will fly!  We're  
looking for speaker presentations for 2009's conference - do you have  
something you want to talk about?  Submit your talk proposal today -  
the proposal window closes on June 1.  AstriCon is the single largest  
conference dedicated to Asterisk applications and business, and this  
year we hope to have even more technical, business, and industry- 
oriented talks that will be incredibly valuable to participants who  
work with Asterisk in some way.

  Asterisk is now the most-installed PBX in North America, and the  
momentum is growing.  There are new apps, new installations, and new  
layers of programs being added to Asterisk every day, and the  
attendees at AstriCon are there so they can hear about what YOU'RE  
doing.   This year we're scaling back to 4 tracks (from 5 last year)  
because too many people missed talks that they had wanted to attend,  
so this means higher attendance for each talk as well as a bit more  
analysis for submitted talk topics.  We already have a good handful of  
speaker topics so far, but it would be great to have another surplus

  Pass this message to your business associates who work with Asterisk  
- it would be great to see some of the more far-flung developers,  
businesses, and users of Asterisk have a chance to participate in the  
larger community of people who make up the Asterisk ecosystem.

Name:   AstriCon 2009
Date:   October 13-15, 2009
Location:   Phoenix, AZ
Close date: June 1

http://www.astricon.net/2009/ - general information
http://www.astricon.net/2009/speaking.php - go directly to sign-up!

Sample Topics:
  - Scaling Asterisk: Thousands of Users or Channels
  - Unique Implementations: Lessons Learned, Methods Used
  - Integrating Asterisk in legacy systems
  - Business Lessons for Asterisk resale or consulting
  - Asterisk in Niche Markets
  - VoIP and Asterisk Security
  - Voice Recognition/Voice Synthesis Topics
  - Regulatory Issues with VoIP
  - Asterisk in all layers of the OSI model
  - "How-To" topics for specific difficult tasks

Tips for speakers:

- Developers:  Your talk proposal should describe what your work is,  
who would benefit from it, and the synopsis should clearly indicate  
who would be interested in this talk. Remember that many of the  
participants aren't C coders, but most will have a very good  
understanding of dialplans and call flows.

- Vendors:  We discourage "advertisements" for products.  Talk about  
how your product can be used in conjunction with other tools to solve  
a problem.  Focus on methods, and use concrete examples or case  
studies instead of on product lines.  The more detailed you make your  
submission as to how attendees will benefit from your talk, the more  
likely your talk will be chosen.

- Integrators: If you'd like to discuss specific case studies, please  
mention company names and size/scope of the installation in your  
synopsis.  It will attract attendees who are looking for real-world  
data.

  As is typical, speakers attend the conference at no cost, which  
given some difficulties in travel budgets may be an incentive for  
perhaps more talk proposals than in previous years (though we had 80+  
last year, which was fantastic!)  If you have any questions regarding  
the proposal methods, or about the conference in general, please drop  
me a line and I'll try to answer your questions or forward you to the  
person who directly handles the issues you may have.  Thanks, and I  
hope to see your talk proposal soon!

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] Add Monitor application to call suppresses audio

[Barry L. Kline]

I have an application where we receive calls on an inbound PRI.  After
hours, our Asterisk box dials our answering service on an outbound PRI
and then bridges the caller to the answering service.   The flow looks
like this:

(CALLER)INBOUND_PRI --> CONTEXT --> GOSUB(Incoming) -->
GOSUB(bridge-to-anssrv) --> DIAL(answering_service) -->
OUTBOUND_PRI(service)

This has been working fine for months without so much as a burp.  What I
need to do is record these calls.

If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio.   Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear the callee, and vice versa.

The fully working code looks like this:
1) exten => s,n(place),Verbose(4,Dialing answering service);
2) exten => s,n,Playback(vrec_prompts/this-call-may-be-recorded);
3) exten => s,n,Set(GROUP()=ANSSVC);
4) exten =>
s,n,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
5) exten => s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten => s,n,Goto(s-${DIALSTATUS},1);

If I insert

exten => s,n,Monitor(wav,${CALLFILENAME},m);

before the dial command on line 5, I'm virtually guaranteed that the
call will fail and no audio will be passed.

I'm using Asterisk 1.6.0.9,  LIBPRI 1.4.10,  and DAHDI 2.1.0.4.

Can anyone shine any light on why this problem is occurring?

TIA,

Barry




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Re: [asterisk-users] Proxying from one server to another

[Adrian Marsh]

Hi David,

 

Thanks for the reply. That's pretty much what I've already tried, but
with no luck on the production machines.  In testing it worked, but the
public IPs and single NICs were causing issues (we believe)

So I was looking for a proxy-type solution.

 

Adrian

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 13 May 2009 15:37
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Proxying from one server to another

 

Redirect traffic with iptables like this:

 

Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
NEW_PUBLIC_IP

 

I'm not sure if this will work for SIP. You may need the proxy to change
info in the sip messages between server and client.

 

--Dave

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: Wednesday, May 13, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Proxying from one server to another

 

Hi All,

 

I'm trying to find a software package to do the following sip proxy
work:

 

I've an A*k server A that needs to be decommissioned, from the USA, and
replaced by server B, in the UK. Both servers are on public internet
IPs.

Whilst the client migration happens, I want to divert all the Register
traffic from Server A to Server B to catch any clients still left out
there.

 

Unfortunately, the original Clients were configured with static IPs
instead of DNS names for the SIP Registrar, so I have to proxy Server A
until all the clients have been updated (which might be a long time).

 

Obviously A*k itself wont do this (as far as I know).  I've looked at
siproxyd and party-sip, but with no success so far.

I've also tried using IPtables to redirect at the IP level, but the
public IP ranges seem to stop me from achieving this. It works in my
local-lan testing, but not on the public servers.

 

Any ideas?

 

Thanks,

 

Adrian

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Re: [asterisk-users] Playback to channel using AMI

[Jon Morgan]

That's superb, thanks very much Jim.

J.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: 13 May 2009 15:54
To: Asterisk User MailList
Subject: Re: [asterisk-users] Playback to channel using AMI

Here is the AMI packet I use to do this:

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1330
Variable: CfMC_WhatToPlay=cfmc/song
Variable: CfMC_WhoHear=SIP/GXP280-16-0844e290
ActionID: callE1330
Async: true


And here are the extensions from extensions.conf: (Watch email line wraps)

exten => do_playback,1,Answer()
exten => do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_playback,n,Wait(0.3)
exten => do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten => do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear} &
${PLAYBACKSTATUS})
exten => do_playback,n,Hangup()

exten => do_chanspy,1,Answer()
exten => do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten => do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,Hangup()



What this does is "dial" an extension to play what I want to play and then
bridge in a ChanSpy with whisper so the extension I want to hear the sound
can "listen" in.

This seems to do the trick for me.

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: Jon Morgan 
> Organization: Complete Automotive Solutions
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Wed, 13 May 2009 11:46:04 +0100
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: [asterisk-users] Playback to channel using AMI
> 
> Hi All,
> 
> I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
> file to a channel using the AMI?
> 
> I've had a play and, as there wasn't a Playback command implemented
directly
> in the AMI, I thought about maybe calling an AGI script from the AMI to do
> this but it seems there's no support for executing AGI through the AMI
> either?
> 
> All I found so far was this:
> 
> 
> http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
> 
> After trying to use "Action: AGI" through the AMI I discovered the
> functionality hasn't been included in the release (Invalid Command in the
> response) and wondered why?
> 
> Is there any other way to playback a wav to a channel using AMI?
> 
> Regards,
> 
> Jon Morgan.
> 
> 
> 
> 
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>http://lists.digium.com/mailman/listinfo/asterisk-users



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No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 8.5.329 / Virus Database: 270.12.27/2111 - Release Date: 05/13/09
07:04:00


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Re: [asterisk-users] Sangoma FXS dialmap

[Doug Lytle]

cb wrote:
> I have a Sangoma A400 card with two FXS ports. They work fine,  
> however as I have analog phones connected, I have no way of telling  
> the phone I am done dialing. Pressing # works fine, but then Asterisk  
>   

That's what the digit and response timeouts are for.  I have:

; 
; Set Timeouts
; 

exten => s,n,Set(TIMEOUT(response)=8)
exten => s,n,Set(TIMEOUT(digit)=2)

Doug


-- 
 
Ben Franklin quote:

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


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Re: [asterisk-users] Proxying from one server to another

[Dan Caescu]

I would try with a b2bua. 

Here's a good (imho) example:

 

http://www.b2bua.org/

 

As a second step to take, I would do automatic tftp/http provisioning for
the devices you have (unless you are talking about softphones). This way you
can specify whichever sip server you want for your devices.

 

Dan

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, May 13, 2009 10:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Proxying from one server to another

 

Redirect traffic with iptables like this:

 

Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
NEW_PUBLIC_IP

 

I'm not sure if this will work for SIP. You may need the proxy to change
info in the sip messages between server and client.

 

--Dave

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh
Sent: Wednesday, May 13, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Proxying from one server to another

 

Hi All,

 

I'm trying to find a software package to do the following sip proxy work:

 

I've an A*k server A that needs to be decommissioned, from the USA, and
replaced by server B, in the UK. Both servers are on public internet IPs.

Whilst the client migration happens, I want to divert all the Register
traffic from Server A to Server B to catch any clients still left out there.

 

Unfortunately, the original Clients were configured with static IPs instead
of DNS names for the SIP Registrar, so I have to proxy Server A until all
the clients have been updated (which might be a long time).

 

Obviously A*k itself wont do this (as far as I know).  I've looked at
siproxyd and party-sip, but with no success so far.

I've also tried using IPtables to redirect at the IP level, but the public
IP ranges seem to stop me from achieving this. It works in my local-lan
testing, but not on the public servers.

 

Any ideas?

 

Thanks,

 

Adrian

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Re: [asterisk-users] Playback to channel using AMI

[Jim Dickenson]

Here is the AMI packet I use to do this:

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1330
Variable: CfMC_WhatToPlay=cfmc/song
Variable: CfMC_WhoHear=SIP/GXP280-16-0844e290
ActionID: callE1330
Async: true


And here are the extensions from extensions.conf: (Watch email line wraps)

exten => do_playback,1,Answer()
exten => do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_playback,n,Wait(0.3)
exten => do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten => do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear} &
${PLAYBACKSTATUS})
exten => do_playback,n,Hangup()

exten => do_chanspy,1,Answer()
exten => do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten => do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${CfMC_WhatToPlay} & ${CfMC_WhoHear})
exten => do_chanspy,n,Hangup()



What this does is "dial" an extension to play what I want to play and then
bridge in a ChanSpy with whisper so the extension I want to hear the sound
can "listen" in.

This seems to do the trick for me.

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: Jon Morgan 
> Organization: Complete Automotive Solutions
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Wed, 13 May 2009 11:46:04 +0100
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: [asterisk-users] Playback to channel using AMI
> 
> Hi All,
> 
> I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
> file to a channel using the AMI?
> 
> I've had a play and, as there wasn't a Playback command implemented directly
> in the AMI, I thought about maybe calling an AGI script from the AMI to do
> this but it seems there's no support for executing AGI through the AMI
> either?
> 
> All I found so far was this:
> 
> 
> http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
> 
> After trying to use "Action: AGI" through the AMI I discovered the
> functionality hasn't been included in the release (Invalid Command in the
> response) and wondered why?
> 
> Is there any other way to playback a wav to a channel using AMI?
> 
> Regards,
> 
> Jon Morgan.
> 
> 
> 
> 
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Re: [asterisk-users] Proxying from one server to another

[David Gibbons]

Redirect traffic with iptables like this:

Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to 
NEW_PUBLIC_IP

I'm not sure if this will work for SIP. You may need the proxy to change info 
in the sip messages between server and client.

--Dave


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh
Sent: Wednesday, May 13, 2009 8:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Proxying from one server to another

Hi All,

I'm trying to find a software package to do the following sip proxy work:

I've an A*k server A that needs to be decommissioned, from the USA, and 
replaced by server B, in the UK. Both servers are on public internet IPs.
Whilst the client migration happens, I want to divert all the Register traffic 
from Server A to Server B to catch any clients still left out there.

Unfortunately, the original Clients were configured with static IPs instead of 
DNS names for the SIP Registrar, so I have to proxy Server A until all the 
clients have been updated (which might be a long time).

Obviously A*k itself wont do this (as far as I know).  I've looked at siproxyd 
and party-sip, but with no success so far.
I've also tried using IPtables to redirect at the IP level, but the public IP 
ranges seem to stop me from achieving this. It works in my local-lan testing, 
but not on the public servers.

Any ideas?

Thanks,

Adrian
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[asterisk-users] Sangoma FXS dialmap

[cb]

I have a Sangoma A400 card with two FXS ports. They work fine,  
however as I have analog phones connected, I have no way of telling  
the phone I am done dialing. Pressing # works fine, but then Asterisk  
passes that # over to the POTS line, and about every 5th call, for  
some reason that is causing the call on the POTS line to fail. The  
suspect the trailing # is also going to get in the way of  
transferring to another extension (or any other dial codes that might  
need to be used from these analog phones).

Is their either a way to strip the # off the end of the dialing so it  
isn't passed thru, or is there a place I can specify a dialmap for  
the FXS ports so when my dial pattern is matched it just dials right  
away (or set a dial timeout as right now there doesn't appear to be  
one, it just waits forever for you to finish dialing).

Or is none of this possible with FXS ports on Sangoma cards and I  
should look at getting an external ATA where I can set these things.


-chris




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[asterisk-users] Request for feedback/testing on Multicast RTP Paging

[Joshua Colp]

Hello everyone,

A month ago I took on an issue on the Asterisk issue tracker 
(https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP 
paging.

This is the ability to send audio to phones (the phone must support it) and 
have it played out the speakerphone. Using multicast RTP is great for
this because it does not incur the cost and weight of setting up a potentially 
short call. Depending on the setup this can actually get to be quite
a big problem because when you involve phones subscribed to the state of 
another they get told that the phone is in use. The amount of SIP traffic can
just spiral out of control.

Originally this issue was filed with a new application that performed the 
paging. I took this application and turned it into a channel driver. This means
that instead of having a dedicated paging application for it you can just use 
Dial(). This also means that in mixed environments you can use the Page()
application along with other phones that do not support the multicast RTP 
paging.

So far I have gotten very little response on the issue so I am asking anyone on 
this mailing list who is interested and has the time to test to please test
and provide some feedback.

A branch based off of trunk (as that is where the channel driver will go) is 
available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797

The dial string for the channel driver is in the form of 
MulticastRTP/// where type is either basic 
or linksys. The
control address is only needed for the linksys type.

Any feedback is welcome as a note on 
https://issues.asterisk.org/view.php?id=11797 and will help to getting this 
into the tree.

Thanks!

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

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Re: [asterisk-users] Switchvox

[Danny Nicholas]

You appear to be quite correct on your Google analysis.  From my light
reading of Digium's description of Switchvox, it is pretty much "Asterisk
for Dummies"; the sole interface and maintenance of the system is a web
interface.  The concept apparently is that Digium provides a running system
"Out-of-the-box" and you aren't allowed to do anything to it except basic
user, queue and extension maintenance.

You could put in a second box just to handle the Polycom stuff if needed or
even do a VM since it would be a SIP-only install;  but then you probably
know more about that than I do.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, May 13, 2009 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Switchvox


I just inherited a client that is using a Switchvox system.  I normally 
install a CentOS based system with freePBX and some custom endpoint 
management stuff for Polycom phones.  This Switchvox is making me feel a 
bit stifled.  I am having nightmares of another recent encounter with 
Trixbox Pro.

Can I really not ssh into this box?  If I could is there anything useful 
that I might change without breaking things and/or endangering their 
warranty or support?  Google seems to be very quiet about customer 
experiences with Switchvox.

j

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Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice router

[Jon Schøpzinsky]

I used wireshark to debug the problem, and I can see that the cisco equipment 
is correctly sending t.38 packets to asterisk, and the whole re-invite process 
is successful.
The problem is, that Asterisk discards the t.38 packets with the error message 
I sent, and therefore the T.38 session never gets underway. Asterisk is stuck 
on the same SEQ id, as it never receives anything from the cisco.
Ive also checked that this isn't a network issue. The packets are coming 
through, asterisk just throws them away with the error message I described.



Med venlig hilsen/Kind Regards

Jon Leren Schøpzinsky
Systems Architect

Firstcom A/S
Bådehavnsgade 2C, 2.
2450 København SV

Web:  http://www.firstcom.dk

-Oprindelig meddelelse-
Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne af David Backeberg
Sendt: 13. maj 2009 14:12
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice 
router

On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky  wrote:
> We are having some problems using t.38 together with a Cisco voice router at
> one of our providers end.
>
> We are using the new digium asterisk fax module to generate the fax, and
> when we use together with our internal Audiocodes Mediant 2000 gateways, we
> have no issues what so ever, and the faxes go right through.
> I can see that asterisk discards all RTP T.38 packets sent from the
> provider, which the error message also indicates.
>
> Is there a known problem, connecting to cisco hardware using t.38 in
> Asterisk 1.6? or does anybody know of a patch that fixes this problem?

I doubt that there is a known problem, as I'm using Cisco with
asterisk and T.38 and having success. Do you have full control over
the Cisco gear?

Please post the dialpeer info from the Cisco gear and I'll take a look
at it. You can also go back through the archives for similar posts
because we've discussed this a few times in the last few months. Among
other things, I saw that your fax tried to transmit at 2400bps. The
gear should be able to support 9600. So that's already fishy.

What happens if you try to send a 'normal' audio fax over voip through
that gear?

 Some things you should know:
* do not compress voip faxes. Faxes are already compressed. If you try
to use a compression codec you'll wreck the fax.
* on the cisco dialpeer be darn sure that you've turned off vad
* for sip on asterisk, you need to enable reinvite, and you also need
to configure a  t38pt_udptl = yes entry in your sip.conf, but you
probably already to that right if you were T.38-ing to the other gear.
Are you sure you weren't just sending a normal audio fax to the other
gear?

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--
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[asterisk-users] Proxying from one server to another

[Adrian Marsh]

Hi All,

 

I'm trying to find a software package to do the following sip proxy
work:

 

I've an A*k server A that needs to be decommissioned, from the USA, and
replaced by server B, in the UK. Both servers are on public internet
IPs.

Whilst the client migration happens, I want to divert all the Register
traffic from Server A to Server B to catch any clients still left out
there.

 

Unfortunately, the original Clients were configured with static IPs
instead of DNS names for the SIP Registrar, so I have to proxy Server A
until all the clients have been updated (which might be a long time).

 

Obviously A*k itself wont do this (as far as I know).  I've looked at
siproxyd and party-sip, but with no success so far.

I've also tried using IPtables to redirect at the IP level, but the
public IP ranges seem to stop me from achieving this. It works in my
local-lan testing, but not on the public servers.

 

Any ideas?

 

Thanks,

 

Adrian

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Re: [asterisk-users] enum agi interesting problem

[Dan Caescu]

That’s not entirely true.
I am using astcc.agi which does exactly this (actually is DeadAGI): dials
the call, and when call is finished, control is given back to the agi script
(for updating cdr and billing the call). 

What I am trying to do is just add a small portion of code in the 'trytrunk'
function so that I can also dial using enum. So far it is partially working:
call gets dialed, but I don't get a status back , so I cannot bill the call
(because I won't know when the call fails since no status is returned).

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Maciejewski
Sent: Wednesday, May 13, 2009 2:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] enum agi interesting problem

Maybe it is something to do with AGI - Dial command.
IFAIK you can't control Dial via AGI script.

>From http://www.voip-info.org/wiki/view/Asterisk+AGI :

Dialing out

If the AGI application dials outward by executing Dial, control over
the call returns to the dialplan and the script loses contact with the
Asterisk server. The script continues to run in the background by
itself and is free to clean up and do post-dial processing.

If you want your application to initiate a call out without being
started through the dialplan:
* Asterisk auto-dial out Move (not copy) a file into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command

Regards,
Chris


2009/5/13 Dan Caescu :
> Forget the typo (s/ANSWERED/ANSWER/g)
>
>
>
> 
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Caescu
> Sent: Tuesday, May 12, 2009 7:07 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] enum agi interesting problem
>
>
>
> Hi,
>
>
>
> I am having a strange problem with enum and AGI.
>
>
>
> Here is what happens:
>
>
>
> I have in my agi something like that:
>
>
>
> foreach my $resolver ("e164.arpa", "e164.info", "e164.org") {
>
>     my @enums = get_enums($phone, $resolver);
>
>     foreach my $enum (@enums) {
>
>     $dialstring = $enum . "|90|HL(" .
> ($maxtime * 60 * 1000) . ":6:3)";
>
>     $res = $AGI->exec("DIAL $dialstring");
>
>     $answeredtime =
> $AGI->get_variable("ANSWEREDTIME");
>
>
>
>     $dialstatus =
> $AGI->get_variable("DIALSTATUS");
>
>     print LOGFILE "Dialstring: $dialstr
> DIALSTATUS: $dialstatus\n";
>
>
>
>     $callstart = time();
>
>     if ($dialstatus eq "ANSWERED") { last;
}
>
>     }
>
>     }
>
> }
>
>
>
> Here’s the output from my logfile:
>
>
>
> Call 1:
>
>
>
> Dialstring:
> sip/16416418003569...@tollfree.sip-happens.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418003569...@sip.tollfreegateway.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring: sip/18003569...@tf.voipmich.com|90|HL(576:6:3)
> DIALSTATUS: ANSWER
>
>
>
> Call 2:
>
>
>
> Dialstring: sip/18002662...@tf.voipmich.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418002662...@sip.tollfreegateway.com|90|HL(576:6:3)
> DIALSTATUS:
>
> Dialstring:
> sip/16416418002662...@tollfree.sip-happens.com|90|HL(576:6:3)
> DIALSTATUS: ANSWER
>
>
>
> And so on.
>
> The call gets answered the first time (call 1 – through sip-happens, call
2,
> through voipmich).
>
> Problem is that after I hang up , it doesn’t return a status, so it cycles
> through the loop and dials the rest of the entries. The last one gets
> dialstatus.
>
>
>
> I believe it’s a stupid mistake but I cannot think of anything right now.
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
> Dan
>
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[asterisk-users] Switchvox

[Jeff LaCoursiere]

I just inherited a client that is using a Switchvox system.  I normally 
install a CentOS based system with freePBX and some custom endpoint 
management stuff for Polycom phones.  This Switchvox is making me feel a 
bit stifled.  I am having nightmares of another recent encounter with 
Trixbox Pro.

Can I really not ssh into this box?  If I could is there anything useful 
that I might change without breaking things and/or endangering their 
warranty or support?  Google seems to be very quiet about customer 
experiences with Switchvox.

j

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Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Cisco voice router

[David Backeberg]

On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky  wrote:
> We are having some problems using t.38 together with a Cisco voice router at
> one of our providers end.
>
> We are using the new digium asterisk fax module to generate the fax, and
> when we use together with our internal Audiocodes Mediant 2000 gateways, we
> have no issues what so ever, and the faxes go right through.
> I can see that asterisk discards all RTP T.38 packets sent from the
> provider, which the error message also indicates.
>
> Is there a known problem, connecting to cisco hardware using t.38 in
> Asterisk 1.6? or does anybody know of a patch that fixes this problem?

I doubt that there is a known problem, as I'm using Cisco with
asterisk and T.38 and having success. Do you have full control over
the Cisco gear?

Please post the dialpeer info from the Cisco gear and I'll take a look
at it. You can also go back through the archives for similar posts
because we've discussed this a few times in the last few months. Among
other things, I saw that your fax tried to transmit at 2400bps. The
gear should be able to support 9600. So that's already fishy.

What happens if you try to send a 'normal' audio fax over voip through
that gear?

 Some things you should know:
* do not compress voip faxes. Faxes are already compressed. If you try
to use a compression codec you'll wreck the fax.
* on the cisco dialpeer be darn sure that you've turned off vad
* for sip on asterisk, you need to enable reinvite, and you also need
to configure a  t38pt_udptl = yes entry in your sip.conf, but you
probably already to that right if you were T.38-ing to the other gear.
Are you sure you weren't just sending a normal audio fax to the other
gear?

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Re: [asterisk-users] Free Fax for asterisk

[David Backeberg]

On Wed, May 13, 2009 at 7:39 AM, Markus Weiler
 wrote:
> I wasn`t very conviced about Spandsp, after trying several versions it
> worked, but not well.

spandsp has been revised to the point that it's now at 0.0.6pre11,
released this month. I've had quite the opposite experience, that
newer versions are working better than previous versions for me.

> We are sending faxes via SIP. When sending faxes from our 1.6 Asterisk
> to our 1.4 Asterisk 50%+ Faxes failed.
> T.38 worked once then stopped I never found the right
> Asterisk-App_fax-Spandsp-dialplan setup again.
> I event patched the app_fax, which contained errors, just to get it working.
> Although I have to admit I installed it on our other Asterisk
> 1.4.17-BRIstuffed (Spandsp 0.0.4-test6) and it works just fine.

You don't say what any of the errors were, so it's hard to tell
whether you encountered bugs or just had configuration issues.

As to fax over SIP, you cannot use a compressed voip codec. It must be
G.711 and you must send faxes no faster than 9600bps. You should also
definitely turn off vad on any intervening voip appliances.

With the 1.6.0 series, regular app_fax, T.38, and the 0.0.6 SpanDSP
series I'm receiving near 99% faxes from real business customers.

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Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold

[Richard Brady]

Hi folks

I am still thinking about the best way to fit this into the config
files, but in the meantime I would like to offer some additional info
in support of my argument for both signalling hold and sending MOH
media. This is quoted from the SIPConnect recommendation from The SIP
Forum, an industry group working towards standardisation:



When the hold initiator (which may be the SIP-PBX or Service Provider
network acting transparently as Media Endpoint) provides music-on-hold
(MOH) treatment:

-   The MOH source in the SIP-PBX or SP-SSE is based on local policy.
-   The hold initiator MUST set the SDP directionality attribute to 
"a=sendonly".

If the hold initiator does not provide MOH, it MUST set the SDP
directionality attribute to "a=inactive" or "a=sendonly". The
attribute "a=inactive" is RECOMMENDED because it provides an
indication to the held entity that MOH is not being provided by the
hold initiator.



For more info see http://www.sipforum.org/sipconnect

Regards,
Richard

On Fri, Apr 3, 2009 at 11:16 AM, Richard Brady  wrote:
> Agreed Olle, it would definitely have to be option driven, not least
> for backward compatibility.
>
> When you say "old idea", is there any discussion we can refer to?
>
> Exisiting variables include:
>
> mohinterpret
> mohsuggest
> musicclass
> musiconhold
>
> The first step would be to clarify what each of these are for. Then
> perhaps we can add options for those which cover the scenarios we are
> interested in.
>
> Of course we we need to understand those scenarios too. So, let's look
> at that. For each channel in the call you need to know how it holds
> and how it likes to be held.
>
> Ways it may hold:
> 1.1. a=sendonly and sends its own MOH (most likely a PBX)
> 1.2. a= sendonly and expects MOH to be generated upstream (most likely
> a handset)
> 1.3. a=inactive and expects MOH to be generated upstream (could be PBX
> or handset)
> 1.4. No signalling, it will simply substitute media
>
> Ways it may like to be held:
> 2.1. Send it a=sendonly and send it MOH (could be PBX or handset)
> 2.2. Send it a= sendonly and no media (inside a network as you mentioned)
> 2.3. Send it a=inactive and no media (could be PBX or handset)
> 2.4. No signalling, simply send it substituted media.
>
> At first glance you would think that it would hold as it likes to be
> held. But actually a handset could use 1.2. while expecting 2.4 as it
> cannot generate hold music for either it's own user when put on hold
> or the remote user when holding.
>
> So do we need two variables with 4 values each? I don't think so. We
> only need to disambiguate between 1.1 and 1.2, and to choose between
> 2.1 through 2.4. Hopefully there is some scope to narrow that down
> further. I will think about it some more.
>
> Giving chan_sip support for the mohinterpret=passthrough option would
> would be a start. But that option itself is ambiguous: does it mean
> media passthrough or signalling passthrough? This ambiguity is
> highlighted in the unanswered message from exvito on this list in
> March last year:
>
> [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical 
> ?
>
> So some thought definitely needs to go into this before it becomes a
> feature request.
>
> R.
>
>
> On Fri, Apr 3, 2009 at 9:03 AM, Olle E. Johansson  wrote:
>> My old idea was to implement an option, since there are many people
>> with different opinions
>> on how a PBX should behave when a channel is put on hold.
>>
>> An option could control how we should handle the bridged channel when
>> the caller or the callee
>> puts a call on hold. It could either be local hold, meaning we
>> entertain the user with music,
>> or a remote hold, which means that we send the hold forward over ISDN
>> or SIP and let the
>> other end handle the hold. This would also work well in larger
>> Asterisk installations,
>> where you don't want to fill up trunks between Asterisk servers with
>> music. The edge server
>> provides the music, no one else.
>>
>> In SIP we could easily add a proprietary header for music class
>> suggestion in these cases.
>>
>> Asterisk admins should be able to set this option per call in the
>> dialplan or per device in
>> channel configurations - or per PBX, also in channel configs.
>>
>> "local hold" or "remote hold" might mean something else, coming to
>> think of it. But it fitted
>> in nicely here :-)
>>
>> /Olle
>>
>> 2 apr 2009 kl. 15.05 skrev Richard Brady:
>>
>>> Furthermore, the following two IETF documents address the need to both
>>> signal the hold and provide the music:
>>>
>>> 1. RFC 5359 (Session Initiation Protocol Service Examples)
>>>
>>> 2. draft-worley-service-example-03 (Session Initiation Protocol
>>> Service Example -- Music on Hold)
>>>
>>> Unfortunately they both address more complex scenarios and solutions,
>>> but they do back me up on the fact that there are good reasons to both
>>> signal hold and provide music.
>>>
>>> R.
>>>
>>> On Wed, Apr 1, 2009 at 6:16 

Re: [asterisk-users] Free Fax for asterisk

[David Backeberg]

On Wed, May 13, 2009 at 3:43 AM, Markus Weiler
 wrote:
> I installed Digiums Free Fax for Asterisk and found out, that it
> automatically retries failed faxes, is there a way to stop that?

You already claimed that this isn't actually the case. I will tell you
that the hardware fax appliances I have used, commonly called "fax
machines" will retry a fax three times before giving up, at least if
they get a busy as opposed to a different kind of failure.

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Re: [asterisk-users] Free Fax for asterisk

[Markus Weiler]

Hi,
it was was my fault, there is no retry ... sorry to bother you.

@David:
I wasn`t very conviced about Spandsp, after trying several versions it 
worked, but not well.
We are sending faxes via SIP. When sending faxes from our 1.6 Asterisk 
to our 1.4 Asterisk 50%+ Faxes failed.
T.38 worked once then stopped I never found the right 
Asterisk-App_fax-Spandsp-dialplan setup again.
I event patched the app_fax, which contained errors, just to get it working.
Although I have to admit I installed it on our other Asterisk 
1.4.17-BRIstuffed (Spandsp 0.0.4-test6) and it works just fine.

I just tried Digiums solution to test if it´s better and it is, all the 
testfaxes went through.
T.38 worked instantly.
Configuration was pretty easy and well documented.
I think $50 per channel is not too much money either, just the 
support...well there is none.

hope i could help

Markus



David Klaverstyn wrote:
> Hi All,
>
> Sorry to hijack this post but I am confused.  What is the advantage of using 
> this "Digium Fax For Asterisk" product when you can use Asterisks' 1.6.x 
> module app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and 
> app_rxfax modules?
>
> Regards
> David.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler
> Sent: Wednesday, 13 May 2009 5:44 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Free Fax for asterisk
>
> Hi,
>
> I installed Digiums Free Fax for Asterisk and found out, that it 
> automatically retries failed faxes, is there a way to stop that?
>
> Thanks
>
> Markus
>
>
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[asterisk-users] Playback to channel using AMI

[Jon Morgan]

Hi All,

I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
file to a channel using the AMI?

I've had a play and, as there wasn't a Playback command implemented directly
in the AMI, I thought about maybe calling an AGI script from the AMI to do
this but it seems there's no support for executing AGI through the AMI
either?

All I found so far was this:


http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/

After trying to use "Action: AGI" through the AMI I discovered the
functionality hasn't been included in the release (Invalid Command in the
response) and wondered why?

Is there any other way to playback a wav to a channel using AMI?

Regards,

Jon Morgan.




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Re: [asterisk-users] Free Fax for asterisk

[David Klaverstyn]

Hi All,

Sorry to hijack this post but I am confused.  What is the advantage of using 
this "Digium Fax For Asterisk" product when you can use Asterisks' 1.6.x module 
app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and app_rxfax 
modules?

Regards
David.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler
Sent: Wednesday, 13 May 2009 5:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Free Fax for asterisk

Hi,

I installed Digiums Free Fax for Asterisk and found out, that it 
automatically retries failed faxes, is there a way to stop that?

Thanks

Markus


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[asterisk-users] AGI scripts in Groovy, JavaScript, JRuby or PHP running on the Java Virtual Machine

[Stefan Reuter]

Hi,

We've just finished adding support for writing AGI scripts in a variety
of popular scripting languages to Asterisk-Java.
The FastAGI server in Asterisk-Java allows you to move your AGI scripts
to a dedicated server and increases performance by eleminating the need
to start the language interpreter for each request.
Our current snapshot release includes an AGI demo in Groovy, JavaScript
and PHP to get you up and running quickly.
Check it out at
http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html
and provide feedback.
Will this also be useful for non-Java developers?

Best regards,

Stefan

-- 
reuter network consulting
Neusser Str. 110
50670 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  stefan.reu...@reucon.com
Jabber:  stefan.reu...@reucon.com
WWW: http://www.reucon.com

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



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Re: [asterisk-users] no source on cdr logs in some cases!!

[Oguzhan Kayhan]

And also
this is the macro for failover

[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1)
exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)
exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} >
2]?${CID_${CALLERID(num)}}:)})
exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1)
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} >
6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})})
exten = s,n,Goto(1-dial,1)
exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
exten = 1-setgbobname,n,Goto(s,3)
exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
exten = 1-fmsetcid,n,Goto(1-dial,1)
exten = 1-dial,1,Dial(${ARG1})
exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1)
exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
exten = 1-CHANUNAVAIL,n,Hangup()
exten = 1-CONGESTION,1,Dial(${ARG2})
exten = 1-CONGESTION,n,Hangup()
exten = 1-out,1,Hangup()


>
> Hello,
> I was experiencing a problem like not seeing source info or caller id on
> some calls.
> When i make a little reserach i figured out that if my dialin plan is like
> this:
> "exten = _00.,1,Macro(trunkdial-failover-0.3
> ${span_1}/9${EXTEN:0},,span_1,span_1)"
>
> I see no source on logfiles.. Sure i have the other fields.
>
> If i change the rule to "exten => _00.,1,Dial(DAHDI/g1/9${EXTEN})"
> everythings goes normal.
> Whats the difference between them(except macro command for sure) or what
> can cause such problem
>
>
> PS: I also have the exact source info in Channel tab of the database as
> SIP/
>
>
>
>
>
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[asterisk-users] Free Fax for asterisk

[Markus Weiler]

Hi,

I installed Digiums Free Fax for Asterisk and found out, that it 
automatically retries failed faxes, is there a way to stop that?

Thanks

Markus


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[asterisk-users] Asterisk 1.6 T.38 generation towards a Cisco voice router

[Jon Schøpzinsky]

Hello List.

 

We are having some problems using t.38 together with a Cisco voice router at 
one of our providers end.

We are using the new digium asterisk fax module to generate the fax, and when 
we use together with our internal Audiocodes Mediant 2000 gateways, we have no 
issues what so ever, and the faxes go right through.

 

When we send faxes to our other provider, who has cisco hardware at their end, 
we get this error:

 

-- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0', [ 006.607923 ], 
STAT_EVT_HW_CLOSE  st: WT_HW_CLSrt: WCLSNCLS

-- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0', [ 006.608118 ], 
STAT_SES_COMPLETE

-- Channel 'SIP/xxx.xx.se-08aaf470' fax session '0' is complete, result: 
'SUCCESS' (FAX_NO_FAX), error: 'CANCELED', pages: 0, resolution: 'unknown', 
transfer rate: '2400', remoteSID: ''

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short 

[Apr 23 15:30:33] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short

 

I can see that asterisk discards all RTP T.38 packets sent from the provider, 
which the error message also indicates.

Is there a known problem, connecting to cisco hardware using t.38 in Asterisk 
1.6? or does anybody know of a patch that fixes this problem?

 

I can see that in the end of the T.38 packet, cisco adds 4 zero fields, which 
are not in the packets that Asterisk sends. Is this some weird 
"we-are-cisco-and-therefore-decide-how-the-packets-should-look"?

 

 

Kind Regards

 

Jon Leren Schøpzinsky

Systems Architect

 

Firstcom A/S

Bådehavnsgade 2C, 2.

2450 København SV

 

Web:  http://www.firstcom.dk  

 

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