james.texter@cox.net
2006-Jan-30 12:42 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I have: group=2 language=en context=from-pstn signalling=fxs_ks channel=>25 I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: -- Called g2/5148346 -- Zap/25-1 answered SIP/412-9b72 However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated.
C F
2006-Jan-31 15:50 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
You could insert a pause by adding a w before the number to be dialed, like this: Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote:> I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: > > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > > span=2,0,0,d4,ami > fxsks=25 > > And in zapata.conf, I have: > group=2 > language=en > context=from-pstn > signalling=fxs_ks > channel=>25 > > I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: > -- Called g2/5148346 > -- Zap/25-1 answered SIP/412-9b72 > > However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
james.texter@cox.net
2006-Feb-01 08:16 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? Thanks again, James ------------------------------ You could insert a pause by adding a w before the number to be dialed, like this: Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote:> > I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: > > > > span=1,1,0,esf,b8zs > > bchan=1-23 > > dchan=24 > > > > span=2,0,0,d4,ami > > fxsks=25 > > > > And in zapata.conf, I have: > > group=2 > > language=en > > context=from-pstn > > signalling=fxs_ks > > channel=>25 > > > > I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: > > -- Called g2/5148346 > > -- Zap/25-1 answered SIP/412-9b72 > > > > However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
C F
2006-Feb-01 09:13 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
Looks like channel 25 is not the one hooked up to your POTS, when an incoming call arrives, what channel does the CLI report? On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote:> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? > > Thanks again, > James > > ------------------------------ > You could insert a pause by adding a w before the number to be dialed, > like this: > Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. > > > On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote: > > > I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: > > > > > > span=1,1,0,esf,b8zs > > > bchan=1-23 > > > dchan=24 > > > > > > span=2,0,0,d4,ami > > > fxsks=25 > > > > > > And in zapata.conf, I have: > > > group=2 > > > language=en > > > context=from-pstn > > > signalling=fxs_ks > > > channel=>25 > > > > > > I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: > > > -- Called g2/5148346 > > > -- Zap/25-1 answered SIP/412-9b72 > > > > > > However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
james.texter@cox.net
2006-Feb-01 10:21 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
The output from the CLI when I put in an inbound call is the following:
-- Starting simple switch on 'Zap/25-1'
-- Executing GotoIf("Zap/25-1",
"1?from-pstn-reghours|s|1:") in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("Zap/25-1",
"0?from-pstn-reghours-nofax|s|1:2") in new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer("Zap/25-1", "") in new stack
-- Executing Wait("Zap/25-1", "1") in new stack
-- Executing SetVar("Zap/25-1", "intype=EXT-412") in new
stack
-- Executing Cut("Zap/25-1", "intype=intype|-|1") in new
stack
It then goes on to call the extension I have setup. I think it's coming in
on Channel 25, but I'm not sure what the -1 is for in Zap/25-1.
Not sure if this is relevant or not, but I'm using a Carrier Access
Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog line
is definitely hooked to the FXO card, and I definitely have the T1 plugged in to
the FXO card.
Thanks,
James
C F wrote:> Looks like channel 25 is not the one hooked up to your POTS, when an
> incoming call arrives, what channel does the CLI report?
>
>
> On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote:
>> Thanks for the reply. I have tried adding anywhere between 1 and 6
w's to the dial string, but still no luck. I hooked up and listened on the
line when the call went out, and never heard any DTMF's. I'm sure this
must be something simple, I just can't seem to figure out for the life of me
what it is. What other information can I provide to help sort this out?
>>
>> Thanks again,
>> James
>>
>> ------------------------------
>> You could insert a pause by adding a w before the number to be dialed,
>> like this:
>> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause.
>>
>>
>> On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote:
>>>> I am experimenting with an asterisk setup in my office. The
last bit I have to test is working with analog lines. I have TE411p digium
card, with an ISDN line plugged into the first, a channel bank plugged into the
second port, and the last two ports empty. I have the following setup in my
zaptel.conf:
>>>>
>>>> span=1,1,0,esf,b8zs
>>>> bchan=1-23
>>>> dchan=24
>>>>
>>>> span=2,0,0,d4,ami
>>>> fxsks=25
>>>>
>>>> And in zapata.conf, I have:
>>>> group=2
>>>> language=en
>>>> context=from-pstn
>>>> signalling=fxs_ks
>>>> channel=>25
>>>>
>>>> I have one analog line plugged in for testing. If I dial that
analog number, the inbound call arrives, and it works great. However, when I
place an outbound call, I get the following output:
>>>> -- Called g2/5148346
>>>> -- Zap/25-1 answered SIP/412-9b72
>>>>
>>>> However, my number never rings. After about 30 seconds, I get
a message saying my call could not be completed as dialed. Almost like it
didn't get all of the digits. Is there a way to inject a pause before
dialing? Any other thoughts? Any help is greatly appreciated.
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
C F
2006-Feb-01 12:31 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
Is this an Adit 600? On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote:> The output from the CLI when I put in an inbound call is the following: > > -- Starting simple switch on 'Zap/25-1' > -- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new stack > -- Goto (from-pstn-reghours,s,1) > -- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack > -- Goto (from-pstn-reghours,s,2) > -- Executing Answer("Zap/25-1", "") in new stack > -- Executing Wait("Zap/25-1", "1") in new stack > -- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack > -- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack > > It then goes on to call the extension I have setup. I think it's coming in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1. > > Not sure if this is relevant or not, but I'm using a Carrier Access Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog line is definitely hooked to the FXO card, and I definitely have the T1 plugged in to the FXO card. > > Thanks, > > James > > > C F wrote: > > Looks like channel 25 is not the one hooked up to your POTS, when an > > incoming call arrives, what channel does the CLI report? > > > > > > On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: > >> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? > >> > >> Thanks again, > >> James > >> > >> ------------------------------ > >> You could insert a pause by adding a w before the number to be dialed, > >> like this: > >> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. > >> > >> > >> On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote: > >>>> I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: > >>>> > >>>> span=1,1,0,esf,b8zs > >>>> bchan=1-23 > >>>> dchan=24 > >>>> > >>>> span=2,0,0,d4,ami > >>>> fxsks=25 > >>>> > >>>> And in zapata.conf, I have: > >>>> group=2 > >>>> language=en > >>>> context=from-pstn > >>>> signalling=fxs_ks > >>>> channel=>25 > >>>> > >>>> I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: > >>>> -- Called g2/5148346 > >>>> -- Zap/25-1 answered SIP/412-9b72 > >>>> > >>>> However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. > >>>> > >>>> _______________________________________________ > >>>> --Bandwidth and Colocation provided by Easynews.com -- > >>>> > >>>> Asterisk-Users mailing list > >>>> To UNSUBSCRIBE or update options visit: > >>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> > >> _______________________________________________ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> Asterisk-Users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> _______________________________________________ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> Asterisk-Users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
james.texter@cox.net
2006-Feb-01 12:50 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
No, it's an Access Bank II SNMP. Thanks, James C F wrote:> Is this an Adit 600? > > On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: >> The output from the CLI when I put in an inbound call is the following: >> >> -- Starting simple switch on 'Zap/25-1' >> -- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new stack >> -- Goto (from-pstn-reghours,s,1) >> -- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack >> -- Goto (from-pstn-reghours,s,2) >> -- Executing Answer("Zap/25-1", "") in new stack >> -- Executing Wait("Zap/25-1", "1") in new stack >> -- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack >> -- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack >> >> It then goes on to call the extension I have setup. I think it's coming in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1. >> >> Not sure if this is relevant or not, but I'm using a Carrier Access Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog line is definitely hooked to the FXO card, and I definitely have the T1 plugged in to the FXO card. >> >> Thanks, >> >> James >> >> >> C F wrote: >>> Looks like channel 25 is not the one hooked up to your POTS, when an >>> incoming call arrives, what channel does the CLI report? >>> >>> >>> On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: >>>> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? >>>> >>>> Thanks again, >>>> James >>>> >>>> ------------------------------ >>>> You could insert a pause by adding a w before the number to be dialed, >>>> like this: >>>> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. >>>> >>>> >>>> On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote: >>>>>> I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: >>>>>> >>>>>> span=1,1,0,esf,b8zs >>>>>> bchan=1-23 >>>>>> dchan=24 >>>>>> >>>>>> span=2,0,0,d4,ami >>>>>> fxsks=25 >>>>>> >>>>>> And in zapata.conf, I have: >>>>>> group=2 >>>>>> language=en >>>>>> context=from-pstn >>>>>> signalling=fxs_ks >>>>>> channel=>25 >>>>>> >>>>>> I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: >>>>>> -- Called g2/5148346 >>>>>> -- Zap/25-1 answered SIP/412-9b72 >>>>>> >>>>>> However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. >>>>>> >>>>>> _______________________________________________ >>>>>> --Bandwidth and Colocation provided by Easynews.com -- >>>>>> >>>>>> Asterisk-Users mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>> >>>> _______________________________________________ >>>> --Bandwidth and Colocation provided by Easynews.com -- >>>> >>>> Asterisk-Users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>>> >>>> _______________________________________________ >>>> --Bandwidth and Colocation provided by Easynews.com -- >>>> >>>> Asterisk-Users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
C F
2006-Feb-01 13:48 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
Then I got no clue how to configure it. But it looks like something is wrong in that setup there. On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote:> No, it's an Access Bank II SNMP. > > Thanks, > > James > > C F wrote: > > Is this an Adit 600? > > > > On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: > >> The output from the CLI when I put in an inbound call is the following: > >> > >> -- Starting simple switch on 'Zap/25-1' > >> -- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new stack > >> -- Goto (from-pstn-reghours,s,1) > >> -- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack > >> -- Goto (from-pstn-reghours,s,2) > >> -- Executing Answer("Zap/25-1", "") in new stack > >> -- Executing Wait("Zap/25-1", "1") in new stack > >> -- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack > >> -- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack > >> > >> It then goes on to call the extension I have setup. I think it's coming in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1. > >> > >> Not sure if this is relevant or not, but I'm using a Carrier Access Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog line is definitely hooked to the FXO card, and I definitely have the T1 plugged in to the FXO card. > >> > >> Thanks, > >> > >> James > >> > >> > >> C F wrote: > >>> Looks like channel 25 is not the one hooked up to your POTS, when an > >>> incoming call arrives, what channel does the CLI report? > >>> > >>> > >>> On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: > >>>> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? > >>>> > >>>> Thanks again, > >>>> James > >>>> > >>>> ------------------------------ > >>>> You could insert a pause by adding a w before the number to be dialed, > >>>> like this: > >>>> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. > >>>> > >>>> > >>>> On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote: > >>>>>> I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: > >>>>>> > >>>>>> span=1,1,0,esf,b8zs > >>>>>> bchan=1-23 > >>>>>> dchan=24 > >>>>>> > >>>>>> span=2,0,0,d4,ami > >>>>>> fxsks=25 > >>>>>> > >>>>>> And in zapata.conf, I have: > >>>>>> group=2 > >>>>>> language=en > >>>>>> context=from-pstn > >>>>>> signalling=fxs_ks > >>>>>> channel=>25 > >>>>>> > >>>>>> I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: > >>>>>> -- Called g2/5148346 > >>>>>> -- Zap/25-1 answered SIP/412-9b72 > >>>>>> > >>>>>> However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. > >>>>>> > >>>>>> _______________________________________________ > >>>>>> --Bandwidth and Colocation provided by Easynews.com -- > >>>>>> > >>>>>> Asterisk-Users mailing list > >>>>>> To UNSUBSCRIBE or update options visit: > >>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>>>> > >>>> _______________________________________________ > >>>> --Bandwidth and Colocation provided by Easynews.com -- > >>>> > >>>> Asterisk-Users mailing list > >>>> To UNSUBSCRIBE or update options visit: > >>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> > >>>> > >>>> _______________________________________________ > >>>> --Bandwidth and Colocation provided by Easynews.com -- > >>>> > >>>> Asterisk-Users mailing list > >>>> To UNSUBSCRIBE or update options visit: > >>>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> > >>> _______________________________________________ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> Asterisk-Users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > >> _______________________________________________ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> Asterisk-Users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
james.texter@cox.net
2006-Feb-01 14:16 UTC
[Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't
I appreciate all of your help. I'm starting to think this is a setup/hardware issue with the channel bank myself. I bought the FXO card off of EBay, so who knows what kind of shape it's in. Does anyone else out here on the forums know how to configure a CAC Access Bank II SNMP channel bank to work with Asterisk? Thanks, James C F wrote:> Then I got no clue how to configure it. But it looks like something is > wrong in that setup there. > > On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: >> No, it's an Access Bank II SNMP. >> >> Thanks, >> >> James >> >> C F wrote: >>> Is this an Adit 600? >>> >>> On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: >>>> The output from the CLI when I put in an inbound call is the following: >>>> >>>> -- Starting simple switch on 'Zap/25-1' >>>> -- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new stack >>>> -- Goto (from-pstn-reghours,s,1) >>>> -- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack >>>> -- Goto (from-pstn-reghours,s,2) >>>> -- Executing Answer("Zap/25-1", "") in new stack >>>> -- Executing Wait("Zap/25-1", "1") in new stack >>>> -- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack >>>> -- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack >>>> >>>> It then goes on to call the extension I have setup. I think it's coming in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1. >>>> >>>> Not sure if this is relevant or not, but I'm using a Carrier Access Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card. The analog line is definitely hooked to the FXO card, and I definitely have the T1 plugged in to the FXO card. >>>> >>>> Thanks, >>>> >>>> James >>>> >>>> >>>> C F wrote: >>>>> Looks like channel 25 is not the one hooked up to your POTS, when an >>>>> incoming call arrives, what channel does the CLI report? >>>>> >>>>> >>>>> On 2/1/06, james.texter@cox.net <james.texter@cox.net> wrote: >>>>>> Thanks for the reply. I have tried adding anywhere between 1 and 6 w's to the dial string, but still no luck. I hooked up and listened on the line when the call went out, and never heard any DTMF's. I'm sure this must be something simple, I just can't seem to figure out for the life of me what it is. What other information can I provide to help sort this out? >>>>>> >>>>>> Thanks again, >>>>>> James >>>>>> >>>>>> ------------------------------ >>>>>> You could insert a pause by adding a w before the number to be dialed, >>>>>> like this: >>>>>> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause. >>>>>> >>>>>> >>>>>> On 1/30/06, james.texter@cox.net <james.texter@cox.net> wrote: >>>>>>>> I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: >>>>>>>> >>>>>>>> span=1,1,0,esf,b8zs >>>>>>>> bchan=1-23 >>>>>>>> dchan=24 >>>>>>>> >>>>>>>> span=2,0,0,d4,ami >>>>>>>> fxsks=25 >>>>>>>> >>>>>>>> And in zapata.conf, I have: >>>>>>>> group=2 >>>>>>>> language=en >>>>>>>> context=from-pstn >>>>>>>> signalling=fxs_ks >>>>>>>> channel=>25 >>>>>>>> >>>>>>>> I have one analog line plugged in for testing. If I dial that analog number, the inbound call arrives, and it works great. However, when I place an outbound call, I get the following output: >>>>>>>> -- Called g2/5148346 >>>>>>>> -- Zap/25-1 answered SIP/412-9b72 >>>>>>>> >>>>>>>> However, my number never rings. After about 30 seconds, I get a message saying my call could not be completed as dialed. Almost like it didn't get all of the digits. Is there a way to inject a pause before dialing? Any other thoughts? Any help is greatly appreciated. >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> --Bandwidth and Colocation provided by Easynews.com -- >>>>>>>> >>>>>>>> Asterisk-Users mailing list >>>>>>>> To UNSUBSCRIBE or update options visit: >>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>>>> >>>>>> _______________________________________________ >>>>>> --Bandwidth and Colocation provided by Easynews.com -- >>>>>> >>>>>> Asterisk-Users mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> --Bandwidth and Colocation provided by Easynews.com -- >>>>>> >>>>>> Asterisk-Users mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>> >>>>> _______________________________________________ >>>>> --Bandwidth and Colocation provided by Easynews.com -- >>>>> >>>>> Asterisk-Users mailing list >>>>> To UNSUBSCRIBE or update options visit: >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>> >>>> _______________________________________________ >>>> --Bandwidth and Colocation provided by Easynews.com -- >>>> >>>> Asterisk-Users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >