[yaman]

i just installed, spa3000, line 1 with broadvoice and pstn line with FWD and i connected my PSTN line into port "line" and my phone into port "phone"

i tested both voip to PSTN and PSTN to voip with and without pin and everything works fine.

i wish to know if there is a possibilty to:

receive a call to FWD and obtain a dialing tone to call internationaly using line 1 account(BV)? in another terms call one voip account and able to receive a dialing tone to call from the other.

i would appreciate a respond with details,


thanks for helping

[mberlant]

This has been asked and answered many times already, most recently by you!

Please read your own thread, http://voxilla.com/forum-viewtopic-t...oiptovoip.html and describe any problems you have understanding the descriptions and responses there. Then, perform your own search for "voip-to-voip" and look through the threads that are listed so that you can understand the configuration considerations necessary to accomplish what you seek.

[yaman]

thanks for your respond, can you explain to me more what do you mean by strap. i could understand that all incoming calls to the account created in line1 will have to go to the FXO port and it will be treated as coming in from PSTN and obtain a dialing tone to enable you dial out using the account created in the PSTN line. so i did conenct the RJ11 cable between FXO & FXS if i understood right i am able to obtain a dialing tone after enabling to yes PSTN>>VOIP but when i dial out the the call does go through only to ON-net calls with a slow motion sound, but for PSTN it does not go through. is there any changes in the setting i need to do for the PSTN line especially in relation to the international control under PSTN line configuration.

thanks

[mberlant]

As explained over there, there is no VoIP-to-VoIP path in the SPA3000. To accomplish this you need to strap (connect with a telephone cord) the PSTN Line jack to the Line 1 jack. This, naturally, disconnects your PSTN service from the SPA-3000.

So, you will dial in to the number associated with your PSTN Line, enter your PIN and be given a Dial Tone to dial out via the service associated with Line 1.

[yaman]

mberlant,

thanks again, actually it is working i have done what you asked me to do but unfortunately i am hearing the receiver voice in slow motion, like his battery is dying.

do you think there should be some changes in the setting?

thanks

[mberlant]

It is possible that your jitter buffer is too big. The jitter buffer stores incoming RTP packets long enough so that the decoding process delivers the voice into your ear in the smoothest way possible. You set its size to balance choppiness against latency. Since you seem to be complaining about excessive latency right now, if your network access quality can tolerate it you may be able to reduce the size of the buffer without introducing noticeable choppiness.

Of course, there's no way to overcome the laws of physics. You don't mention where in the world you are, but if your electrons are traveling a long distance to get where they need to go latency will remain a challenge. For example, my Asterisk is in Japan, which means that most of my connections have 300ms latency. There's just no way to get electrons to move faster than the Speed of Light.

[yaman]

thanks once more for your reply, actually i am located in europe, i did several test and got the following results:

i have three 3 accounts with the same provider so i did register account no1 on my PAP2 to make outbound call, account no2 to the LINE1 of sipura3000 and account no3 with PSTN line of sipura 3000. i have set all the three account to G729a since my bandwidth is max 100Kbps upload and 600Kbps download.

test1: calling from accno1>>>accno2 obtain a bip than dial the pin obtain a dialing tone and make an outbound call. results: slow motion receiving, even you can notice that when you are dialing the destinaton number it takes time to connect.

test2: calling from accno1>>>accno3 obtain and bip than dial the pin, obtain a dialign tone and make an outbound call. results: same as test1

test3: calling from PSTN>>>account2 obtain a bip than dial the pin obtain a dialing tone and make an outbound call. results: the same as test1 & test2

test4: connect a PSTN line into line port of sipura 3000, call the PSTN number obtain a dialing tone and dial out using account2. results: quality of call is ok

NB: when i ping the provider of the 3 above mentioned account, i get max 140ms

thanks

[yaman]

i wish to add also that i did another setting which is as follow:

i registered the PAP2 accountNo1 with an IPBX and register also LINE1 of sipura 3000 into the same IPBX and PSTN line with the voip provider directly without going through the IPBX.


so when i call from accountno1 to accountno2 and obtain a bip and dial the pin and dialing out after getting a dialign tone the call goes through with the same problem as before but if i dial to the extention number of the accountno2 and obtain the dialing tone than the call goes thrugh without a problem,


thanks

[mberlant]

Now I think I understand your problem better. I had believed that you were complaining about voice latency - the time it takes for a word spoken by one person to be heard by the other person. Now it appears that you are really complaining about the time it takes for a dialed call to connect.

If this is true, there is only one element of the puzzle that you can control - the Dial Plan. When you dial a numeric string into your SPA the SPA matches the digits you have dialed so far with the Dial Plan elements programmed into it. The big advantage in a well crafted Dial Plan is if it has discrete elements to handle each unambiguous number string you wish to handle. This will allow the SPA to process a call as soon as you dial the last digit in the sequence because it knows already that for this sequence there are no more digits coming. If you don't do this your SPA must wait 3 seconds longer for the "next" digit that never comes before "giving up" and processing the call.

There are several threads here that describe in detail the logic that needs to go into a good Dial Plan. The best one is a Sticky thread at the top of this forum.

[yaman]

i do not think it is a dial plan issue, because whenever i dial a number i always end it with the "#" key so the call immediately go through, but you hear the dialing in background. the receiver of the call you hear has a slow motion voice. i wonder if we have an issue of bandwidth or not since my upload is always 100Kbps.

the other strange thing is when i do register accountno1 and account no2 on my IPBX and assign them with extention numbers and keep accountno3 with the voip provider directly, when i do call from extention of accountno1>>>extention of accountno2 extention and ontain a bip and a dial tone to call out using account no3 the call goes through without a slow motion voice and it is in good quality.


i was wondering if it is an issue of bandwidth, since when i call from accountno1>>>accountno2 than obtain access to account no3 to dial out i am already on a three leg call on the same bandwidth but i have set all the account on G729a which should not be a problem, but also why also i think it might not be a bandwidth issue, when i call using a PSTN to the accountno2 and obtain access to accountno3 to make a call out i am using in this case two legs and the bandwidth required is less, and still i am getting the slow motion voice of the receiver.

it is a strange thing, can you keep updating me with ideas and your valuable comments

thanks