Google Voice Outbound Trunk: Howto Add Custom Dial String

yeshappytrader · **Joined:** Wed Feb 09, 2011 9:15 pm **Posts:** 16

Hello All,

I am able to make a direct telephone call through Asterisk to Google Voice like this:

exten => 2122222222,1,Dial(gtalk/asterisk/[email protected])

But everytime I try and write that same dial string in a2billing under " Trunks "
" Additional Parameters ", a2billing ALWAYS sends the call out as follows SIP:

Code:

<SIP/2122222222-00000072>AGI Rx << STREAM FILE dollars "#" 0
-- Playing 'dollars' (escape_digits=#) (sample_offset 0)
<SIP/2122222222-00000072>AGI Tx >> 200 result=0 endpos=7200
<SIP/2122222222-00000072>AGI Rx << CHANNEL STATUS SIP/2122222222-00000072
<SIP/2122222222-00000072>AGI Tx >> 200 result=6
<SIP/2122222222-00000072>AGI Rx << GET DATA prepaid-press9-new-speeddial 5000 1
-- <SIP/2122222222-00000072> Playing 'prepaid-press9-new-speeddial.gsm' (language 'en')
<SIP/2122222222-00000072>AGI Tx >> 200 result=1
<SIP/2122222222-00000072>AGI Rx << GET DATA prepaid-enter-dest 6000 20
-- <SIP/2122222222-00000072> Playing 'prepaid-enter-dest.gsm' (language 'en')

JABBER: asterisk INCOMING:
<SIP/2122222222-00000072>AGI Tx >> 200 result=12123333333
<SIP/2122222222-00000072>AGI Rx << EXEC DIAL SIP/Dial(gtalk/asterisk/[email protected]),60,HRrL(2147483647:61000:30000)12123333333@default2
-- AGI Script Executing Application: (DIAL) Options: (SIP/Dial(gtalk/asterisk/[email protected]),60,HRrL(2147483647:61000:30000)12123333333@default2)
> Limit Data for this call:
> timelimit = 2147483647 ms (2147483.647 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
[May 28 09:39:17] ERROR[20138]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("voice.google.com)", "(null)", ...): Name or service not known
[May 28 09:39:17] WARNING[20138]: chan_sip.c:5330 create_addr: No such host: voice.google.com)
[May 28 09:39:17] WARNING[20138]: acl.c:708 ast_ouraddrfor: Cannot connect
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to (null):
INVITE sip:Dial(gtalk/asterisk/[email protected]) SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XXX.XXX:5060;branch=z9hG4bK3a95a70f

How do I get A2Billing to send the call out WITHOUT sending out this string:

DIAL SIP/Dial(gtalk/asterisk/[email protected]) = DOES NOT WORK

I need it to send this string:

Dial(gtalk/asterisk/[email protected]) = DOES WORK

Any assistance would be greatly appreciated.

chrismartin4u · **Joined:** Tue May 08, 2012 6:03 am **Posts:** 2

I think I also would like to know it.

flash0151 · **Joined:** Thu Jul 18, 2013 4:29 am **Posts:** 2

bump i need this as well anyone?

provided by Star2Billing S.L.

Google Voice Outbound Trunk: Howto Add Custom Dial String

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