Xfering PSTN calls (on *1) to VOIP calls to * 2 (openwrt)

[LoneShadow]

Hi,

I have Asterisk running on my linksys router one in US, and another in India.
What is the best way to call my US pstn #, while the call is ringing, asterisk can take a pin #, and let me dial an Indian #. Which would then get routed to my other Asterisk+SPA3K setup. Trying to make my US asterisk box like a calling card.

Something like (rough rules)
exten => s,1, Answer
exten => s,2, Ring
exten => s,3, Wait(1)
.
.
exten to handle pin
.
.
exten => _011N, 7, Dial(IAX/India,30,t)
.
.
; else let the local phone ring for other callers
exten => .., .. , Dial(SIP/Phone,30,t)


My second question is about using a calling card from my sip client. If I dial a 1800 #, asterisk forwards the call to a voip service provider. And them when the calling card # asks for pin or # to dial. If I enter the digits on the sip phone, those numbers dont reach my calling card #. What needs to configured and on sip phone or asterisk ?

Any tips/pointers would be greatly appreciated

Thanks,
LS

[prayer]

I, too am just starting up with asterisk (really astlinux). Similar to yours but runs on OLD PC and has router, QOS , etc. I think you have that too. I probably wont be getting to passing tones to an FXO until a month from now. Sorry I can't help you right now. But this keeps this thread more activated in light that someone will have a good answer for the both of us.
Rich

[LoneShadow]

aah should have updatd the thread
Found the answers after searching some more
1) Use DISA for providing dial tone
2) use dtfm=inband for allowing pressing of #s while on call

Looks like WRT54GS cannot handle inband , if this is the case, then there is no point using DISA. I am trying out Asterisk@Home, once I get that working, will try to use those scripts on openwrt or on my old PII webserver running asterisk.

- LS

[mberlant]

The router has no effect upon the transmission success/failure of DTMF. You need to figure out which of the three DTMF transmission modes your VoIP service provider requires and set that Asterisk interface to use that transmission mode.

[LoneShadow]

Originally Posted by mberlant

The router has no effect upon the transmission success/failure of DTMF. You need to figure out which of the three DTMF transmission modes your VoIP service provider requires and set that Asterisk interface to use that transmission mode.

I was under the impression that linksys ruoter dosnt perform well with dtmf=inband.

I tried playing the demo context when I recieve the call from my sip provider, if dtfm is set to rfc2833. I could hear the demo clearly, but was not able to send any signal for when I pressed a number to choose from the menu.

After doing some searches, I arrived at a conclusion that one needs dtmf set to inband if we need to do IVR.

- LS

[mberlant]

I can't imagine any way for a router to decrypt the payload of an RTP packet, determine that the user is trying to send a DTMF signal inside that packet, and then treat that packet more poorly than any RTP packet that contains only voice. Based upon this logic, I made the statement that the router can't adversely affect DTMF transmission.

[LoneShadow]

Originally Posted by mberlant

I can't imagine any way for a router to decrypt the payload of an RTP packet, determine that the user is trying to send a DTMF signal inside that packet, and then treat that packet more poorly than any RTP packet that contains only voice. Based upon this logic, I made the statement that the router can't adversely affect DTMF transmission.

Hmm, I am a lil confused, and I guess I confused you as well

Asterisk supposedly takes more cpu cycle if its inband mode. Since linksys router is 200Mhz, it seemed to perform badly with asterisk/openwrt firmware on it.

Once I do more research, I guess I will be able to explain my problem in a better way, or find solution on my own

- LS

[mberlant]

I see where the confusion comes from. When you used the word "router" I interpreted that to be the router function, which remains completely independent of the Asterisk function under OpenWRT.

Even with that distinction, though, I have run Asterisk on OpenWRT on WRT54GS v2 hardware on several occasions and have not bumped into that problem. I have noted (and documented here in this forum) that there is enough horsepower to run Asterisk, including several dozen simultaneous (client and service) Registrations and several simultaneous conversations. The number of conversations was limited by WAN bandwidth and not apparantly by WRT horsepower, but there was not enough horsepower to do any voice processing, like IVR or voice mail or CODEC transcoding.

I did not observe a problem with inband DTMF processing, but it is possible that the inband DTMF was encoded in the ATA and not transcoded in Asterisk, but was decoded only at the destination gateway. The last time I actively worked with Asterisk/OpenWRT was several months ago, so my memory is a bit fuzzy now.

[LoneShadow]

Ok I have some more updates and few more questions

I tried my voip provider using thier given adapter directly (without asterisk). I am able to pass dial tones during a call.
I used tcpdump to verify that G711 Ulaw was being used. Not sure dtfmmode is indicated in the trace.

So I transfer the following settings to my asterisk sip profiles
dtmfmode=rfc2833
disallow=all
allow=ulaw

Now I disable the adapter, and use my asterisk instead, and tried the following tests :-

a) from a Cisco SIP phone, I register into my asterisk box, set the context to demo. I can dial into my asterisk, and the demo plays, I can choose all the menus.


b) now I change the context to talk to my voip provider for the Cisco SIP phone. I place a call out to check my voice mails (external). Dial tones dont work.

c) using Xlite, I dial the external voicemail, I can press *, #s, and those go to my voicemail. (Also going thru asterisk and voip provider)

d) Now I change the context for xlite to demo. Dial into asterisk, I hear the demo, I try to choose sometihng from the IVR, and it dosnt work.

--- And this is where I start getting confused again :roll:

I tried tcpdump on the xlite traffic to the external voicemail system. It looked like it had RTP and RTCP packets. My speculation is Xlite uses RTCP packets for sending dial tones, which my asterisk can forward without any issues.

I have not verified if my Cisco phone sends the RTCP packers or not, but it can some how send dialtones during a call only to asterisk.

Anybody have any idea where I am going wrong ?

Thanks,
LS

[chandave]

RTCP are used for status info. X-Lite, by default, will send RTCP packets unless you disable this feature. Look under
Main Menu > Advanced System Settings > RTP Settings > Send RTCP Messages

Is your Cisco phone setup to send DTMF as rfc2833 RTP encoded packets? If not then it might be sending it using Cisco's own DTMF RTP packet. So, if Asterisk is just bridging the call (no protocol conversion), Asterisk will not try to convert the Cisco DTMF RTP to normal DTMF RTP encoding.

Is your X-Lite set for inband DTMF, rfc2833, or AVT/SIP INFO?

See ya...

d.c.