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ozimarco · 12-16-2011, 04:25 PM

Quote:

Originally Posted by Corbu'

Anyone managed to get CallThru working with Tropo ? How is it ?

I am happy to report that, after a few days of trying and with a lot of help from Aaron, I can now answer your question in the affirmative.

I think, if you follow the SIPSorcery recipe and also refer to this forum thread SIP Sorcery Community Forums • View topic - Tropo Callback question , you should breeze through it. Good luck!

Corbu' · 12-16-2011, 08:51 PM

Excellent. Could you please tell me how long after the call is connected you get the dial tone and can dial thru? Is it more than 1-2 seconds? Also, is there any delay in the conversation when using the callthru method compared with a traditional ATA or SIP softphone?

ozimarco · 12-16-2011, 09:25 PM

Quote:

Originally Posted by Corbu'

Excellent. Could you please tell me how long after the call is connected you get the dial tone and can dial thru? Is it more than 1-2 seconds? Also, is there any delay in the conversation when using the callthru method compared with a traditional ATA or SIP softphone?

As soon as the call is connected to Tropo, you are immediately asked to enter your PIN. As soon as you have entered your PIN, you are asked to enter your number and press #. The delay to connect to your number is not more than 1-2 seconds. It all happens quite quickly. I was only testing by calling myself and the delay was minimal, about the same as when you do an echo test. I will try and do some real world testing soon.

Juste · 12-20-2011, 06:00 AM

Hi,

Did you easily cope with setting incoming rules ?
I find the explanations quite confusing and I am still waiting for Aaron's answer to my questions.

ozimarco · 12-20-2011, 06:30 AM

Quote:

Originally Posted by Juste

Hi,

Did you easily cope with setting incoming rules ?
I find the explanations quite confusing and I am still waiting for Aaron's answer to my questions.

By default, to accept incoming calls, you don't need any incoming rules. The only incoming rule I have at the moment is to send calls to CallCentric voicemail if they are not answered in 20 seconds. Before setting up voicemail, I didn't have any rules and could still receive calls. By default, all incoming calls are sent to all your SIP accounts, i.e. if you have more than one.

I have two SIP accounts. To determine which SIP account calls from certain SIP Providers will go to, I click on the SIP Provider and fill in the Contact Details of the SIP account I want to send the incoming calls to, e.g. sip:account1@sipsorcery.com .

If you do have any incoming rules, make sure you select an In Dial Plan in your SIP Account settings, otherwise they won't take effect. Outgoing and incoming dial plans for a SIP account are usually configured on the same dial plan page, so, in my case, I have selected the same dial plan name for Out Dial Plan and In Dial plan.

I agree it's all a bit confusing at first but it does start to make sense after a while.

By the way, Aaron's a busy man, but he does try and answer each and every question you throw at him. I noticed that he usually answers my posts in the evening so he probably has a day job, too, God forbid!

Juste · 12-20-2011, 01:42 PM

Thanks for replying.

Now, reading your message and Aaron's answer I feel even more confused

There must be a way to make those explanations easier-to-understand. And the more I read his guide, the more I feel stuck.
Frustrating.

ozimarco · 12-20-2011, 02:14 PM

What are the incoming rules you are having problems with?

Juste · 12-20-2011, 02:33 PM

I have trouble understanding the formula they use to have incoming calls calls on different providers forwarded to different devices registered with my accounts.

For example, I have three different accounts 123, 124 and 125 linked to three devices at different places.
I also have three providers Provider1, Provider2 and Provider3.
I want incoming calls on Provider1 to be forwarded to account 123, incoming calls on Provider2 to be forwarded to account 124 and incoming calls on Provider3 to be forwarded to account 125.

How can you achieve this ?

ozimarco · 12-20-2011, 02:51 PM

That is too easy! You create 3 SIP accounts: 123@sipsorcery.com, 124@sipsorcery.com and 125@sipsorcery.com .

You create your 3 SIP Providers. Tick the Register box on each provider.
For provider1, enter the Register Contact as 123@sipsorcery.com
For provider2, enter the Register Contact as 124@sipsorcery.com
For provider3, enter the Register Contact as 125@sipsorcery.com

That's all that's needed. No need to enter anything in the In Dial Plan.

Juste · 12-21-2011, 04:16 AM

Thanks for the tip.
It did the trick. I think the instructions rather misled me. Will have to find out the rest step by step.

12-20-2011, 02:14 PM	#47
ozimarco Senior Member Join Date: May 2006 Location: Lower Chittering, about an hour from Perth, Western Australia Posts: 229 Thanks: 61 Thanked 42 Times in 28 Posts	What are the incoming rules you are having problems with? __________________ ISP: Internode Easy Reach 60: 60GB for AUD60 VSPs: Pennytel and others via SIPSorcery Hardware: Billion 7800VDPX + Siemens Gigaset A580IP and Yealink T22P.

12-20-2011, 02:51 PM	#49
ozimarco Senior Member Join Date: May 2006 Location: Lower Chittering, about an hour from Perth, Western Australia Posts: 229 Thanks: 61 Thanked 42 Times in 28 Posts	That is too easy! You create 3 SIP accounts: 123@sipsorcery.com, 124@sipsorcery.com and 125@sipsorcery.com . You create your 3 SIP Providers. Tick the Register box on each provider. For provider1, enter the Register Contact as 123@sipsorcery.com For provider2, enter the Register Contact as 124@sipsorcery.com For provider3, enter the Register Contact as 125@sipsorcery.com That's all that's needed. No need to enter anything in the In Dial Plan. __________________ ISP: Internode Easy Reach 60: 60GB for AUD60 VSPs: Pennytel and others via SIPSorcery Hardware: Billion 7800VDPX + Siemens Gigaset A580IP and Yealink T22P.

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12-16-2011, 08:51 PM	#42
Corbu' Member Join Date: May 2009 Location: Canada Posts: 81 Thanks: 14 Thanked 6 Times in 5 Posts	Excellent. Could you please tell me how long after the call is connected you get the dial tone and can dial thru? Is it more than 1-2 seconds? Also, is there any delay in the conversation when using the callthru method compared with a traditional ATA or SIP softphone?

12-20-2011, 06:00 AM	#44
Juste Member Join Date: Mar 2007 Location: Toronto, Canada Posts: 65 Thanks: 8 Thanked 5 Times in 4 Posts	Hi, Did you easily cope with setting incoming rules ? I find the explanations quite confusing and I am still waiting for Aaron's answer to my questions.

12-20-2011, 01:42 PM	#46
Juste Member Join Date: Mar 2007 Location: Toronto, Canada Posts: 65 Thanks: 8 Thanked 5 Times in 4 Posts	Thanks for replying. Now, reading your message and Aaron's answer I feel even more confused There must be a way to make those explanations easier-to-understand. And the more I read his guide, the more I feel stuck. Frustrating.

12-20-2011, 02:33 PM	#48
Juste Member Join Date: Mar 2007 Location: Toronto, Canada Posts: 65 Thanks: 8 Thanked 5 Times in 4 Posts	I have trouble understanding the formula they use to have incoming calls calls on different providers forwarded to different devices registered with my accounts. For example, I have three different accounts 123, 124 and 125 linked to three devices at different places. I also have three providers Provider1, Provider2 and Provider3. I want incoming calls on Provider1 to be forwarded to account 123, incoming calls on Provider2 to be forwarded to account 124 and incoming calls on Provider3 to be forwarded to account 125. How can you achieve this ?

12-21-2011, 04:16 AM	#50
Juste Member Join Date: Mar 2007 Location: Toronto, Canada Posts: 65 Thanks: 8 Thanked 5 Times in 4 Posts	Thanks for the tip. It did the trick. I think the instructions rather misled me. Will have to find out the rest step by step.