[yaman]

i have two accounts registered with line1 and PSTN line respectively with the same provider.

when i dial the PSTN number i get the bip, so i enter the the pin number followed by # than i get the line1 dilaing tone, while dialing the number i keep hearing the dialing tone, than it disconnect after i finish dialing the number.

can someone tells me what is happening, i am using the firmware 3.1.7(GWg)

in the past 3 month ago, it worked perfectely with firmware 3.1.3

i would appreciate some help

thanks

[yaman]

Originally Posted by yaman

i have two accounts registered with line1 and PSTN line respectively with the same provider.

when i dial the PSTN number i get the bip, so i enter the the pin number followed by # than i get the line1 dilaing tone, while dialing the number i keep hearing the dialing tone, than it disconnect after i finish dialing the number.

can someone tells me what is happening, i am using the firmware 3.1.7(GWg)

in the past 3 month ago, it worked perfectely with firmware 3.1.3

i would appreciate some help

thanks

i wish to add to the above that the above problem is a result of calling from a third voip account with the same provider on the same network.

when i do dial rom a mobile phone to the PSTN number which is connected to the line port of the spa3000, the call goes through without a problem with an echo heard by the receiving party, meanign the final destination who is receiving the call is hearing an echo (is there s olution for the echo problem?)

is it possible that the first problem is a codec issue since the originating call from the third voip account is coming from G711 to the PSTN number connected to the line port than bridging to the voip account on line1 which is set to G711 as well.

thanks for any comment

[jcgalvezv]

Yaman:

If it was working perfectly with firmware version 3.1.3, Why don't you go back to that version? Is there anything new in version 3.1.7 you consider you need? if the answer is no, you don't really need version 3.1.7 and I would recommend you to go back to version 3.1.3.

Juan C.

[yaman]

Originally Posted by jcgalvezv

Yaman:

If it was working perfectly with firmware version 3.1.3, Why don't you go back to that version? Is there anything new in version 3.1.7 you consider you need? if the answer is no, you don't really need version 3.1.7 and I would recommend you to go back to version 3.1.3.

Juan C.

thanks for your worry, actually it is working from mobile phone but not from voip, i will try later on to see if it will work from PSTN to PSTN>>VOIP. if it works than most probably might be an issue with the provider remote gateway which needs to add something to his dial peer.

thanks

[yaman]

Originally Posted by yaman

thanks for your worry, actually it is working from mobile phone but not from voip, i will try later on to see if it will work from PSTN to PSTN>>VOIP. if it works than most probably might be an issue with the provider remote gateway which needs to add something to his dial peer.

thanks

it was exactely as i expected it, so i did added few options to the dial peer and now it can take DTMF.

few additional questions:

1- how can the caller passes the PIN security process by entering his caller id number so the spa3000 can identidy the caller as a friendly one and immediately provide dialing tone without going through the process of entering the pin number

2. if this is possible how many friendly numbers can the spa3000 take by accepting any of these friendly callers and provide them dialing tone.

thanks

[jcgalvezv]

Include your friend's caller id number in PSTN Line tab, PSTN-To-VoIP Gateway Setup section, PSTN Caller ID Pattern field. This field is a comma separeted list of caller id number that will bypass authentication.

Juan C.

[yaman]

thanks,

in the case where the caller is not part of this friendly list, will the call ring on the phone set which is connected to the phone port and no authentification bip will come through?

what i wanted to achieve is the following:

friendly callers will get voip dialing tone and if the caller is not a friendly one than the phone will continue ringing on the analogue set which is connected to the phone port of the SPA without getting the authentification bip. is this possible?

on the other hand, can you explain to me what are these parameters for:

PSTN CID For VoIP CID:

PSTN CID Number Prefix:

thanks

[yaman]

Originally Posted by yaman

thanks,

in the case where the caller is not part of this friendly list, will the call ring on the phone set which is connected to the phone port and no authentification bip will come through?

what i wanted to achieve is the following:

friendly callers will get voip dialing tone and if the caller is not a friendly one than the phone will continue ringing on the analogue set which is connected to the phone port of the SPA without getting the authentification bip. is this possible?

on the other hand, can you explain to me what are these parameters for:

PSTN CID For VoIP CID:

PSTN CID Number Prefix:

thanks

what i wanted to know as well is the following:

if i set the PSTN answer delay to high number that takes the caller 5 ringback before getting the authentification bip and a friendly caller calls the PSTN number, will he gets the dialing tone immediately or after the 5 ringback?

thanks

[jcgalvezv]

After 5 rings.

Juan C.

[yaman]

on the other hand, can you explain to me what are these parameters for:

PSTN CID For VoIP CID:

PSTN CID Number Prefix:

thanks in advance