Re: [asterisk-users] Google Voice
I'm using chan_motif with Asterisk 11. It still works. I actually received an email from google yesterday that there had been no traffic on my number lately so the number would be reclaimed. I had switched my outgoing away from GV several months ago when they were supposed to discontinue the service. I switched back to it yesterday and have made several calls. No problems. On Sat, Jan 17, 2015 at 7:35 AM, CDR vene...@gmail.com wrote: Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice
Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice
On Mon, Nov 17, 2014 at 6:37 PM, George Wu aihu...@gmail.com wrote: anybody know the motif driver if the integration with google voice still work or not? What's the best way for the interop with google voice? https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google is the relevant documentation on the wiki. I haven't tried it myself in a long while, however Google was supposed to end XMPP support for GV back in May. I've heard mixed reports from community members. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] google voice
anybody know the motif driver if the integration with google voice still work or not? What's the best way for the interop with google voice? Thanks. George wu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Calls Fail
A quick update. The nick: theory was proven to be wrong. The incoming calls consistently fail with or without nick: tag. I am concentrating on the incoming calls for now. -Vladimir On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote: Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the nick: structure in the failed request. Outgoing calls connect intermittently, but no sound path gets established. Any ideas? Thank you, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Calls Fail
If anybody reads this thread here is the solution. It appeared to be some strange corruption of my Asterisk. As I started debugging and recompiled everything returned back to normal. What still puzzles me how some Google Voice accounts continued working all the time. -Vladimir On 7/22/2013 12:02 PM, Vladimir Mikhelson wrote: A quick update. The nick: theory was proven to be wrong. The incoming calls consistently fail with or without nick: tag. I am concentrating on the incoming calls for now. -Vladimir On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote: Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the nick: structure in the failed request. Outgoing calls connect intermittently, but no sound path gets established. Any ideas? Thank you, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice Calls Fail
Hi All: Has anybody tackled the latest Google Voice issue where incoming and outgoing calls for certain Google Voice accounts fail? I have filed the bug report with details https://issues.asterisk.org/jira/browse/ASTERISK-22176 For incoming calls Asterisk does not reply to the initial XML request coming from Google Voice. Detailed comparison to a successful call initiation shows the lack of the nick: structure in the failed request. Outgoing calls connect intermittently, but no sound path gets established. Any ideas? Thank you, Vladimir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice with Asterisk 11/chan_motif
Dear Mr. Colp and/or anyone who can help, Recently Ive upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I dont have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Thanks in advance. Best regards, Josué Freitas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Josue Freitas wrote: Dear Mr. Colp and/or anyone who can help, Recently I’ve upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I don’t have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Google is responsible for sending the call to you. If you get nothing on your screen after executing xmpp set debug on and placing a call to your Google Voice number then Google is not sending the call to you. You can try restarting Asterisk to see if that makes it work. There's nothing that can be done to force them to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, February 05, 2013 7:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif Josue Freitas wrote: Dear Mr. Colp and/or anyone who can help, Recently I've upgraded to Asterisk 11 and setup chan_motif for Google Voice. Outbound calls are working good but I don't have any inbound traffic through GV. I did all I could find on Google but nothing solved. GV settings seems to be right (Google Chat is enabled with no voicemail access). I always receive calls when on Gmail but when I close the browser no activity happens on Asterisk (xmpp set debug on). BTW, I have no traffic at all on XMPP port (5222). Is that really needed to have GV working with Asterisk/chan_motif? Google is responsible for sending the call to you. If you get nothing on your screen after executing xmpp set debug on and placing a call to your Google Voice number then Google is not sending the call to you. You can try restarting Asterisk to see if that makes it work. There's nothing that can be done to force them to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
Might also want to check the google hasnt detected an unusual login and is asking for the ip to be accepted. Log in to gmail with that account and check Sent from my iPhone 5 On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote: Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif
I indeed access Gmail and GV from a different IP than the Asterisk server, but just made it from there and it's ok. The Asterisk server is in the US but I'm currently abroad. Is that a problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert-GMAIL Sent: Tuesday, February 05, 2013 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif Might also want to check the google hasnt detected an unusual login and is asking for the ip to be accepted. Log in to gmail with that account and check Sent from my iPhone 5 On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote: Josue Freitas wrote: Thank you! What about the XMPP traffic? Even when I place calls using GV there's no XMPP traffic on 5222. Do I really need to have the XMPP port (5222) open in the firewall? Asterisk acts as an XMPP client. It establishes an outgoing connection to port 5222 of the Google Talk XMPP server. No incoming connections occur. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com actually resolves to a
range of IP addresses. When you set up your firewall, it may not be
including all of the possible resolutions for voice.google.com...
voice.l.google.com. 300 IN A 74.125.225.36
voice.l.google.com. 300 IN A 74.125.225.46
voice.l.google.com. 300 IN A 74.125.225.33
voice.l.google.com. 300 IN A 74.125.225.32
voice.l.google.com. 300 IN A 74.125.225.41
voice.l.google.com. 300 IN A 74.125.225.38
voice.l.google.com. 300 IN A 74.125.225.35
voice.l.google.com. 300 IN A 74.125.225.39
voice.l.google.com. 300 IN A 74.125.225.40
voice.l.google.com. 300 IN A 74.125.225.34
voice.l.google.com. 300 IN A 74.125.225.37
(ie 74.125.225.32-41 and 74.125.225.46)
Since these are short TTL values (the 300 means 5 minutes) there may be a
brief period where your devices and your firewall agree, before one or both
change their mind about the IP address behind that hostname.
I just tried out of the blue calling from D70 through Google Voice to a
cell phone, and it worked. I hung up, redial, and no audio at all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from voice.google.com
10,000:20,000 from my Airport Express public IP and from voice.google.com
My issue is that when I place a call with google voice, I have no audio
path at all in both way.
When a call is received on google voice (and sent to the D70), if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are setup as
follow:
;Dialing out on google voice:
exten = _1zxxzxx,1,Dial(Gtalk/**asterisk/+${EXTEN}@voice.**
google.com exten...@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com,1,Verbose(0, Incoming gtalk from
${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private networks.
Thanks a ton for the help !
--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
__**__**_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Google voice with no voice
Chris,
I covered the whole 74.125.225.* subnet.
Even if I open the ports mentioned below for all (not limited to IP
addresses) I still have the same issue.
Have anyone ever succeeded in such configuration? :
Digium phones on 2 different private networks (2 different buildings)
Asterisk server in the internet with a public IP
Use Google Voice
Even if you have asterisk on a private network, but have the same kind
of solution working for you, I'd love to hear your story..
On 1/22/13 9:55 AM, Christopher Harrington wrote:
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com
mailto:fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com
http://voice.google.com actually resolves to a range of IP addresses.
When you set up your firewall, it may not be including all of the
possible resolutions for voice.google.com...
voice.l.google.com http://voice.l.google.com.300INA74.125.225.36
voice.l.google.com http://voice.l.google.com.300INA74.125.225.46
voice.l.google.com http://voice.l.google.com.300INA74.125.225.33
voice.l.google.com http://voice.l.google.com.300INA74.125.225.32
voice.l.google.com http://voice.l.google.com.300INA74.125.225.41
voice.l.google.com http://voice.l.google.com.300INA74.125.225.38
voice.l.google.com http://voice.l.google.com.300INA74.125.225.35
voice.l.google.com http://voice.l.google.com.300INA74.125.225.39
voice.l.google.com http://voice.l.google.com.300INA74.125.225.40
voice.l.google.com http://voice.l.google.com.300INA74.125.225.34
voice.l.google.com http://voice.l.google.com.300INA74.125.225.37
(ie 74.125.225.32-41 and 74.125.225.46)
Since these are short TTL values (the 300 means 5 minutes) there may be
a brief period where your devices and your firewall agree, before one or
both change their mind about the IP address behind that hostname.
I just tried out of the blue calling from D70 through Google Voice
to a cell phone, and it worked. I hung up, redial, and no audio at all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try
Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and
connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from
voice.google.com http://voice.google.com
10,000:20,000 from my Airport Express public IP and from
voice.google.com http://voice.google.com
My issue is that when I place a call with google voice, I have
no audio
path at all in both way.
When a call is received on google voice (and sent to the D70),
if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are
setup as
follow:
;Dialing out on google voice:
exten =
_1zxxzxx,1,Dial(Gtalk/__asterisk/+${EXTEN}@voice.__google.com
mailto:exten...@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com mailto:r...@gmail.com,1,Verbose(0,
Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the
audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private
networks.
Thanks a ton for the help !
--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
You are obviously getting the call connected, so the subnet issue is moot.
What this sounds like (pardon the pun) to me is an rtp skip issue. The
working calls are generating rtp connections in the allowed range; the
other calls have one or more ports outside of your rtp range. Verify that
all of your ports defined in rtp.conf (1-2 by default) are open in
the firewall.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Tuesday, January 22, 2013 10:18 AM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Google voice with no voice
Chris,
I covered the whole 74.125.225.* subnet.
Even if I open the ports mentioned below for all (not limited to IP
addresses) I still have the same issue.
Have anyone ever succeeded in such configuration? :
Digium phones on 2 different private networks (2 different buildings)
Asterisk server in the internet with a public IP Use Google Voice
Even if you have asterisk on a private network, but have the same kind of
solution working for you, I'd love to hear your story..
On 1/22/13 9:55 AM, Christopher Harrington wrote:
On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com
mailto:fr...@efirehouse.com wrote:
Actually, the funny thing is that it works randomly.
This may be due to the fact that voice.google.com
http://voice.google.com actually resolves to a range of IP addresses.
When you set up your firewall, it may not be including all of the
possible resolutions for voice.google.com...
voice.l.google.com http://voice.l.google.com.300INA74.125.225.36
voice.l.google.com http://voice.l.google.com.300INA74.125.225.46
voice.l.google.com http://voice.l.google.com.300INA74.125.225.33
voice.l.google.com http://voice.l.google.com.300INA74.125.225.32
voice.l.google.com http://voice.l.google.com.300INA74.125.225.41
voice.l.google.com http://voice.l.google.com.300INA74.125.225.38
voice.l.google.com http://voice.l.google.com.300INA74.125.225.35
voice.l.google.com http://voice.l.google.com.300INA74.125.225.39
voice.l.google.com http://voice.l.google.com.300INA74.125.225.40
voice.l.google.com http://voice.l.google.com.300INA74.125.225.34
voice.l.google.com http://voice.l.google.com.300INA74.125.225.37
(ie 74.125.225.32-41 and 74.125.225.46)
Since these are short TTL values (the 300 means 5 minutes) there may be
a brief period where your devices and your firewall agree, before one or
both change their mind about the IP address behind that hostname.
I just tried out of the blue calling from D70 through Google Voice
to a cell phone, and it worked. I hung up, redial, and no audio at
all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try
Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and
connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express
--
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from
voice.google.com http://voice.google.com
10,000:20,000 from my Airport Express public IP and from
voice.google.com http://voice.google.com
My issue is that when I place a call with google voice, I have
no audio
path at all in both way.
When a call is received on google voice (and sent to the D70),
if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are
setup as
follow:
;Dialing out on google voice:
exten =
_1zxxzxx,1,Dial(Gtalk/__asterisk/+${EXTEN}@voice.__google.com
mailto:exten...@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com mailto:r...@gmail.com,1,Verbose(0,
Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate
Re: [asterisk-users] Google voice with no voice
Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect to google voice using gtalk. Here is my physical configuration: Digium D70 -- private network 192.168.1.x -- Airport express -- Internet -- Asterisk with public IP My asterisk has the following ports open: 5060 tcp/udp from my Airport Express public IP and from voice.google.com http://voice.google.com 10,000:20,000 from my Airport Express public IP and from voice.google.com http://voice.google.com My issue is that when I place a call with google voice, I have no audio path at all in both way. When a call
Re: [asterisk-users] Google voice with no voice
Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect to google voice using gtalk. Here is my physical configuration: Digium D70 -- private network 192.168.1.x -- Airport express -- Internet -- Asterisk with public IP My asterisk has
Re: [asterisk-users] Google voice with no voice
Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect
Re: [asterisk-users] Google voice with no voice
Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http://voice.l.google.com.300INA74.125.225.32 voice.l.google.com http://voice.l.google.com.300INA74.125.225.41 voice.l.google.com http://voice.l.google.com.300INA74.125.225.38 voice.l.google.com http://voice.l.google.com.300INA74.125.225.35 voice.l.google.com http://voice.l.google.com.300INA74.125.225.39 voice.l.google.com http://voice.l.google.com.300INA74.125.225.40 voice.l.google.com http://voice.l.google.com.300INA74.125.225.34 voice.l.google.com http://voice.l.google.com.300INA74.125.225.37 (ie
Re: [asterisk-users] Google voice with no voice
If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com http://voice.l.google.com.300INA74.125.225.36 voice.l.google.com http://voice.l.google.com.300INA74.125.225.46 voice.l.google.com http://voice.l.google.com.300INA74.125.225.33 voice.l.google.com http
Re: [asterisk-users] Google voice with no voice
Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Actually, the funny thing is that it works randomly. This may be due to the fact that voice.google.com http://voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com
Re: [asterisk-users] Google voice with no voice
Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open the ports mentioned below for all (not limited to IP addresses) I still have the same issue. Have anyone ever succeeded in such configuration? : Digium phones on 2 different private networks (2 different buildings) Asterisk server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network
Re: [asterisk-users] Google voice with no voice
*CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more ports outside of your rtp range. Verify that all of your ports defined in rtp.conf (1-2 by default) are open in the firewall. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial
Re: [asterisk-users] Google voice with no voice
What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls are generating rtp connections in the allowed range; the other calls have one or more
Re: [asterisk-users] Google voice with no voice
*CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI *CLI -- SIP/D70-0006 is ringing *CLI -- SIP/D70-0006 answered Gtalk/+xx-2310 == Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited non-zero on 'Gtalk/+xx-2310' On 1/22/13 11:21 AM, Danny Nicholas wrote: You are obviously getting the call connected, so the subnet issue is moot. What this sounds like (pardon the pun) to me is an rtp skip issue. The working calls
Re: [asterisk-users] Google voice with no voice
This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice configuration and have the calls routed to my Google Chat only, this is what happens: The Asterisk receives the call. The D70 rings. If I pick up, nothing happens (I see on the D70 display that I picked up) The caller still hear the ringing tone THat's what I see on the console: *CLI -- Executing [r...@gmail.com@gtalk_incoming:1] Verbose(Gtalk/+1xx-2310, 0, Incoming gtalk from +1xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= ) in new stack Incoming gtalk from +xxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU= -- Executing [r...@gmail.com@gtalk_incoming:2] Answer(Gtalk/+xx-2310, ) in new stack -- Executing [r...@gmail.com@gtalk_incoming:3] Wait(Gtalk/+xx-2310, 2) in new stack -- Executing [r...@gmail.com@gtalk_incoming:4] Dial(Gtalk/+xx-2310, SIP/D70) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/D70 *CLI
Re: [asterisk-users] Google voice with no voice
That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice Danny, Thanks for the trick, that made all outgoing calls working. Now, the issue is with incoming calls. Even if I turn off all other phones in google voice
Re: [asterisk-users] Google voice with no voice
This sounds like a codec issue. Set your verbose to 10 and retry the incoming call. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-2 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has CONNECTED status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: Do a netstat -anp during the call. This will (hopefully) show you where the out of range condition
Re: [asterisk-users] Google voice with no voice
OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked up.. I could not hear anything on the D70.. But if I push 1 (which is the google voice option to pickup the screened call), then the audio path works in both way. So the real issue is that when google voice talks when I pick up to let me know who's calling, I can't hear anything, until I press a digit. If I press 1, I get the call connected. If I press 2, I can hear google voice. The question is why can't I hear google voice right away without pushing a digit ? I tried to go into google voice configuration and remove the call screening, but it looks like for calls on gtalk , the screening is always active. So I guess I will know that I need to press 1 or 2 from the D70 for everything to work. It slightly sucks, but I'll take it. On 1/22/13 2:29 PM, Danny Nicholas wrote: This sounds like a codec issue. Set your verbose to 10 and retry the incoming call. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1x@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI -- SIP/D70-0004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI core show help gtalk gtalk show channels Show GoogleTalk channels *CLI gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Ohai from Asterisk timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re: [asterisk-users] Google voice with no voice I added port 5061 without success. I am wondering if I used a man in the middle like iptel.org service, it would work ? On 1/22/13 12:00 PM, Danny Nicholas wrote: Each asterisk call uses 3 ports; 5060 is used
Re: [asterisk-users] Google voice with no voice
Frank wrote: OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked up.. I could not hear anything on the D70.. But if I push 1 (which is the google voice option to pickup the screened call), then the audio path works in both way. So the real issue is that when google voice talks when I pick up to let me know who's calling, I can't hear anything, until I press a digit. If I press 1, I get the call connected. If I press 2, I can hear google voice. The question is why can't I hear google voice right away without pushing a digit ? This is a Google Voice thing. Even the Google talk client itself sends a digit of 1 when you answer the call. That being said you can do this from inside of Asterisk dialplan with a combination of Answer, Wait, and SendDTMF(1) -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
Hi , So I tried Answer() Wait(1) SendDTMF(1) But I got an error in the console: [Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4) If I do core show application sendDTMF , nothing comes up. If there anything special to compile for this ? Thanks On 1/22/13 2:54 PM, Joshua Colp wrote: Frank wrote: OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked up.. I could not hear anything on the D70.. But if I push 1 (which is the google voice option to pickup the screened call), then the audio path works in both way. So the real issue is that when google voice talks when I pick up to let me know who's calling, I can't hear anything, until I press a digit. If I press 1, I get the call connected. If I press 2, I can hear google voice. The question is why can't I hear google voice right away without pushing a digit ? This is a Google Voice thing. Even the Google talk client itself sends a digit of 1 when you answer the call. That being said you can do this from inside of Asterisk dialplan with a combination of Answer, Wait, and SendDTMF(1) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
Frank wrote: Hi , So I tried Answer() Wait(1) SendDTMF(1) But I got an error in the console: [Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4) The app_senddtmf.so module has to be built and loaded. You can load it explicitly using module load app_senddtmf.so. If that fails then it was not built and you will have to look into why not. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
My bad, I found it not loaded in my modules.conf. This is now working. What a pain. Is there a wiki page I can update in order to share the configuration and how to have this work, with everybody ? On 1/22/13 2:58 PM, Joshua Colp wrote: Frank wrote: Hi , So I tried Answer() Wait(1) SendDTMF(1) But I got an error in the console: [Jan 22 14:54:13] WARNING[28067]: pbx.c:4458 pbx_extension_helper: No application 'SendDTMF' for extension (gtalk_incoming, r...@gmail.com, 4) The app_senddtmf.so module has to be built and loaded. You can load it explicitly using module load app_senddtmf.so. If that fails then it was not built and you will have to look into why not. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google voice with no voice
Frank wrote: My bad, I found it not loaded in my modules.conf. This is now working. What a pain. Is there a wiki page I can update in order to share the configuration and how to have this work, with everybody ? A wiki page for using it with the unsupported chan_gtalk / res_jabber combination is available at: https://wiki.asterisk.org/wiki/display/AST/Old+Calling+using+Google A new channel driver for Asterisk 11 called chan_motif was written which replaces chan_gtalk and is fully supported. Details on using it with Google Voice is available at: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google voice with no voice
Greetings all,
I was reading the documentation tonight, and decided to try Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from voice.google.com
10,000:20,000 from my Airport Express public IP and from voice.google.com
My issue is that when I place a call with google voice, I have no audio
path at all in both way.
When a call is received on google voice (and sent to the D70), if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are setup as
follow:
;Dialing out on google voice:
exten = _1zxxzxx,1,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com,1,Verbose(0, Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private networks.
Thanks a ton for the help !
--
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Re: [asterisk-users] Google voice with no voice
Actually, the funny thing is that it works randomly.
I just tried out of the blue calling from D70 through Google Voice to a
cell phone, and it worked. I hung up, redial, and no audio at all.
On 1/21/13 10:38 PM, Frank wrote:
Greetings all,
I was reading the documentation tonight, and decided to try Google voice
with my asterisk.
I was able to setup iksemel, connect to google using jabber, and connect
to google voice using gtalk.
Here is my physical configuration:
Digium D70 -- private network 192.168.1.x -- Airport express --
Internet -- Asterisk with public IP
My asterisk has the following ports open:
5060 tcp/udp from my Airport Express public IP and from voice.google.com
10,000:20,000 from my Airport Express public IP and from voice.google.com
My issue is that when I place a call with google voice, I have no audio
path at all in both way.
When a call is received on google voice (and sent to the D70), if I pick
up, nothing happen, and the caller still hear the ringing tone.
My D70 is setup as follow in the sip.conf:
[D70]
type=friend
nat=yes
qualify=yes
directmedia=no
host=dynamic
secret=takapoum
disallow=all
allow=ulaw
context=LocalSets
mailbox=D70@default
my gtalk.conf is setup as follow:
[general]
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=gtalk_incoming
connection=asterisk
and finally, the interesting parts in my extensions.conf are setup as
follow:
;Dialing out on google voice:
exten = _1zxxzxx,1,Dial(Gtalk/asterisk/+${EXTEN}@voice.google.com)
same = n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten = r...@gmail.com,1,Verbose(0, Incoming gtalk from ${CALLERID(all)})
same = n,Answer()
same = n,Wait(2)
same = n,Dial(SIP/D70)
same = Hangup()
I would appreciate if anyone could give me a hint about the audio path.
This is a project that we I will try to setup in a small fire
department, and before I try it, I would like to make sure that my
Digium phones will be able to get full audio path behind private networks.
Thanks a ton for the help !
--
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Re: [asterisk-users] Google voice with no voice
On 1/21/2013 7:59 PM, Frank wrote: Actually, the funny thing is that it works randomly. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. In the past, I have had strange behaviors like this as well. Turned out to be a ARP race condition with my firewall with static IP assignments. As soon as the second device would ARP, I would loose connectivity with the first device. Check that you have no other device using the IP address that your D70 is using. Also, make sure that nothing else is competing with the Google Voice registration. -- Jim Lucas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
Chris Datfung wrote: Hi, Hola, I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Are there additional parts to your configuration files? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com wrote: I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. Are there additional parts to your configuration files? I ran make examples after I installed asterisk, so the rest of the configuration files are what ever defaults are normally created. Thanks, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
Chris Datfung wrote: On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someone help me figure out why incoming Google voice calls are not ringing on the Iaxy? Did chan_motif successfully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. You can manually load it using module load chan_motif.so and it will say if it has been loaded or the error if it could not be loaded. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
On Mon, Nov 26, 2012 at 3:53 PM, Joshua Colp jc...@digium.com wrote: Chris Datfung wrote: On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: Hi Joshua, How can I verify that chan_motif successfully loaded? I didn't see any errors during the build process. You can manually load it using module load chan_motif.so and it will say if it has been loaded or the error if it could not be loaded. Hi Joshua, I can confirm that chan_motif succesfully loaded: asterisk*CLI module load chan_motif.so Unable to load module chan_motif.so Command 'module load chan_motif.so' failed. [Nov 26 09:04:33] WARNING[28686]: loader.c:868 load_resource: Module 'chan_motif.so' already exists. I restarted Asterisk but Google Voice calls are still not forwarded to my iaxy. Any other ideas how to debug this? Thanks, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
Chris Datfung wrote: Hi Joshua, I can confirm that chan_motif succesfully loaded: asterisk*CLI module load chan_motif.so Unable to load module chan_motif.so Command 'module load chan_motif.so' failed. [Nov 26 09:04:33] WARNING[28686]: loader.c:868 load_resource: Module 'chan_motif.so' already exists. I restarted Asterisk but Google Voice calls are still not forwarded to my iaxy. Any other ideas how to debug this? Nothing else immediately springs to mind I'm afraid. Everything looks as though it should be working and I've checked the code to make sure the session initiation is proper. I'll see if I can reproduce this over the next few days in my spare time. To others using chan_motif - are you experiencing the same issue? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google Voice seems to be broken now. Call is received, but Asterisk does nothing: --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=078099D69B89C046 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-initiate sid=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:jin=urn:xmpp:jingle:1jin:content name=audio creator=initiatorrtp:description media=audio ssrc=731587560 xmlns:rtp=urn:xmpp:jingle:apps:rtp:1rtp:payload-type id=103 name=ISAC clockrate=16000/rtp:payload-type id=104 name=ISAC clockrate=32000/rtp:payload-type id=107 name=speex clockrate=16000rtp:parameter name=bitrate value=22000//rtp:payload-typertp:payload-type id=9 name=G722 clockrate=16000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=102 name=ILBC clockrate=8000rtp:parameter name=bitrate value=13300//rtp:payload-typertp:payload-type id=108 name=speex clockrate=8000rtp:parameter name=bitrate value=11000//rtp: - --- XMPP received from 'google-cathy' --- payload-typertp:payload-type id=0 name=PCMU clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=8 name=PCMA clockrate=8000rtp:parameter name=bitrate value=64000//rtp:payload-typertp:payload-type id=127 name=red clockrate=8000/rtp:payload-type id=126 name=telephone-event clockrate=8000/rtp:rtcp-mux/rtp:encryptionrtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2//rtp:encryption/rtp:descriptionp:transport xmlns:p=http://www.google.com/transport/p2p//jin:content/jin:jingleses:session type=initiate id=c1654741541 initiator= - --- XMPP received from 'google-cathy' --- jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=103 name=ISAC clockrate=16000/pho:payload-type id=104 name=ISAC clockrate=32000/pho:payload-type id=107 name=speex bitrate=22000 clockrate=16000/pho:payload-type id=9 name=G722 bitrate=64000 clockrate=16000/pho:payload-type id=102 name=ILBC bitrate=13300 clockrate=8000/pho:payload-type id=108 name=speex bitrate=11000 clockrate=8000/pho:payload-type id=0 name=PCMU bitrate=64000 clockrate=8000/pho:payload-type id=8 name=PCMA bitrate=64000 clockrate=8000/pho:payload-type id=127 name=red clockrate=8000/pho:payload-type id=126 name=telephone-event clockrate=8000/pho:rtcp-mux/pho:src-id731587560/pho:src-idrtp:encryption xmlns:rtp= - --- XMPP received from 'google-cathy' --- urn:xmpp:jingle:apps:rtp:1rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_80 key-params=inline:t/ni1bJ62BAh0CYQgH0LebZabWx47cG7iou0/OsJ tag=1/rtp:crypto crypto-suite=AES_CM_128_HMAC_SHA1_32 key-params=inline:ysx82SVYw1H61YGmaV2d0b32zxvRBtf6PvBMlhwR tag=2/pho:usage//rtp:encryption/pho:description/ses:session/iq - --- XMPP received from 'google-cathy' --- iq type=set to=cathy.fou...@gmail.com/asterisk-x217D1B44 id=7B548BACBF5495D3 from=jeandenis.gir...@gmail.com/gmail.3027C461jin:jingle action=session-terminate sid=c1654741541 xmlns:jin=urn:xmpp:jingle:1ses:reason xmlns:ses=http://www.google.com/session;ses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//jin:jingleses:session type=terminate id=c1654741541 initiator=jeandenis.gir...@gmail.com/gmail.3027C461 xmlns:ses=http://www.google.com/session;ses:reasonses:connectivity-error//ses:reasonpho:call-ended xmlns:pho=http://www.google.com/session/phone//ses:session/iq - Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEUEARECAAYFAlCzs60ACgkQuu7Rv+oOo/iAvQCYlWFMToLIl3CFtYLhCCpQBbZx WACeJ6xBAn1c/JU+U7kqqlvAZvPr+lk= =DOBH -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice Routing
Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google Voice seems to be broken now. stripped The signaling you've posted isn't actually from Google Voice, it's from Google Talk. While they both go through the Google XMPP server the signaling is far far different. Just right now I tested both a Gmail client calling into Asterisk and Google Voice calling into Asterisk. Both are working as expected for me. This narrows things down to the following: 1. Configuration issue as has been discussed for both of you 2. Google Talk client changes that chan_motif isn't tolerant of yet 3. Google Voice gateway changes (limited to some) that chan_motif isn't tolerant of yet It's probably #1 _ but I have nothing to immediately suggest, I'll keep thinking and looking. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice and back (chan_motif)
Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Hola, Today I started to experiment with Google Voice and Asterisk-11.0.1. Awesome! Following the instructions on the wiki (https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was able to make / receive calls quite easily with a single account on asterisk. Then I tried to add a second Google Voice account to Asterisk, and make calls between accounts. I defined a second connection in xmpp.conf, a second account in chan_motif (see relevant configuration below). I'm getting the following error: ERROR[28651][C-0002]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session (see full log below) Should I open a bug report or did I make an mistake in configuration? You've found a bug! I've fixed it now, though. It'll go out in the next Asterisk 11 release or you can check out Asterisk 11 from subversion to get it. The issue in question is that the candidates were indeed incomplete according to the specification because we were not putting a network attribute within them. I've fixed this so we do and also made the ICE-UDP candidate interpretation code that output the message above more forgiving, specifically it no longer requires them. They are for debugging purposes and aren't used in chan_motif. This didn't show up earlier since many clients just don't require it, and Google Talk/Google Voice don't use ICE-UDP candidates. Sorry for the inconvenience! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 06/11/2012 02:16, Joshua Colp a écrit : You've found a bug! I've fixed it now, though. It'll go out in the next Asterisk 11 release or you can check out Asterisk 11 from subversion to get it. I have applied the patch, it now works as I expected: I can make calls from sip phone1 connected to Asterisk, through my Google Voice account to another Google Voice account, and receive on sip phone2, connected to the same Asterisk. Awesome! Sorry for the inconvenience! No problem Joshua, thanks for very prompt fix! Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCZOpMACgkQuu7Rv+oOo/jJdwCaAyw+unmXEpH8vHYBQiiBDe4z 9ygAnjNQKFmuUvMdLnv7/sblJNr0k5oW =V11X -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Today I started to experiment with Google Voice and Asterisk-11.0.1. Following the instructions on the wiki (https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was able to make / receive calls quite easily with a single account on asterisk. Then I tried to add a second Google Voice account to Asterisk, and make calls between accounts. I defined a second connection in xmpp.conf, a second account in chan_motif (see relevant configuration below). I'm getting the following error: ERROR[28651][C-0002]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session (see full log below) Should I open a bug report or did I make an mistake in configuration? motif.conf: - --- [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-cathy ; - xmpp.conf xmpp.conf: - -- [google-jd] type=client serverhost=talk.google.com username=jeandenis.gir...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 [google-cathy] type=client serverhost=talk.google.com username=cathy.fou...@gmail.com secret= priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 extensions.conf: - [incoming-motif] exten = s,1,NoOp() same = n,Wait(1) same = n,Answer() same = n,SendDTMF(1) same = n,Dial(SIP/FYJmmzJ3,20) call log: - - == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [72@i9PuqEcv:1] Dial(SIP/i9PuqEcv-0002, Motif/google-jd/cathy.fou...@gmail.com,,r) in new stack --- XMPP sent to 'google-jd' --- iq from='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' to='cathy.fou...@gmail.com/asterisk-xD2C13566' type='set' id='o'jingle action='session-initiate' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1' initiator='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06'content creator='initiator' name='audio'description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'payload-type id='110' name='speex' channels='1' clockrate='8000'/payload-type id='0' name='PCMU' channels='1' clockrate='8000'/payload-type id='9' name='G722' channels='1' clockrate='8000'/payload-type id='8' name='PCMA' channels='1' clockrate='8000'/payload-type id='101' name='telephone-event' channels='1' clockrate='8000'//descriptiontransport xmlns='urn:xmpp:jingle:transports:ice-udp:1'//content/jingle/iq - -- Called Motif/google-jd/cathy.fou...@gmail.com --- XMPP received from 'google-jd' --- - --- XMPP received from 'google-cathy' --- iq from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 type=set id=ojingle action=session-initiate sid=7e44df781ce623b6 initiator=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 xmlns=urn:xmpp:jingle:1content creator=initiator name=audiodescription media=audio xmlns=urn:xmpp:jingle:apps:rtp:1payload-type id=110 name=speex channels=1 clockrate=8000/payload-type id=0 name=PCMU channels=1 clockrate=8000/payload-type id=9 name=G722 channels=1 clockrate=8000/payload-type id=8 name=PCMA channels=1 clockrate=8000/payload-type id=101 name=telephone-event channels=1 clockrate=8000//descriptiontransport xmlns=urn:xmpp:jingle:transports:ice-udp:1//content/jingle/iq - --- XMPP sent to 'google-cathy' --- iq type='result' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='o'/ - --- XMPP sent to 'google-cathy' --- iq from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' type='set' id='j'jingle action='transport-info' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1'content creator='responder' name='audio'transport xmlns='urn:xmpp:jingle:transports:ice-udp:1' pwd='4b4001b575f3c7b824e14d9436d5f466' ufrag='6c28e0a07a5269e82ee313d916a046f7'candidate component='1' foundation='583375015' generation='0' id='0a86' ip='192.168.1.1' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='583378294' generation='0' id='3c7f' ip='192.168.0.10' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='192809686' generation='0' id='85cc' ip='123.50.122.114' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='2' foundation='583375015' generation='0' id='cc6e' ip='192.168.1.1' port='16385' priority='2130706430' protocol='udp' type='host'/candidate component='2' foundation='583378294' generation='0' id='5cb8' ip='192.168.0.10' port='16385'
Re: [asterisk-users] Google Voice and back (chan_motif)
Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf On 11/05/2012 08:35 PM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Today I started to experiment with Google Voice and Asterisk-11.0.1. Following the instructions on the wiki (https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was able to make / receive calls quite easily with a single account on asterisk. Then I tried to add a second Google Voice account to Asterisk, and make calls between accounts. I defined a second connection in xmpp.conf, a second account in chan_motif (see relevant configuration below). I'm getting the following error: ERROR[28651][C-0002]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session (see full log below) Should I open a bug report or did I make an mistake in configuration? motif.conf: - --- [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-cathy ; - xmpp.conf xmpp.conf: - -- [google-jd] type=client serverhost=talk.google.com username=jeandenis.gir...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 [google-cathy] type=client serverhost=talk.google.com username=cathy.fou...@gmail.com secret= priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 extensions.conf: - [incoming-motif] exten = s,1,NoOp() same = n,Wait(1) same = n,Answer() same = n,SendDTMF(1) same = n,Dial(SIP/FYJmmzJ3,20) call log: - - == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [72@i9PuqEcv:1] Dial(SIP/i9PuqEcv-0002, Motif/google-jd/cathy.fou...@gmail.com,,r) in new stack --- XMPP sent to 'google-jd' --- iq from='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' to='cathy.fou...@gmail.com/asterisk-xD2C13566' type='set' id='o'jingle action='session-initiate' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1' initiator='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06'content creator='initiator' name='audio'description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'payload-type id='110' name='speex' channels='1' clockrate='8000'/payload-type id='0' name='PCMU' channels='1' clockrate='8000'/payload-type id='9' name='G722' channels='1' clockrate='8000'/payload-type id='8' name='PCMA' channels='1' clockrate='8000'/payload-type id='101' name='telephone-event' channels='1' clockrate='8000'//descriptiontransport xmlns='urn:xmpp:jingle:transports:ice-udp:1'//content/jingle/iq - -- Called Motif/google-jd/cathy.fou...@gmail.com --- XMPP received from 'google-jd' --- - --- XMPP received from 'google-cathy' --- iq from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 type=set id=ojingle action=session-initiate sid=7e44df781ce623b6 initiator=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 xmlns=urn:xmpp:jingle:1content creator=initiator name=audiodescription media=audio xmlns=urn:xmpp:jingle:apps:rtp:1payload-type id=110 name=speex channels=1 clockrate=8000/payload-type id=0 name=PCMU channels=1 clockrate=8000/payload-type id=9 name=G722 channels=1 clockrate=8000/payload-type id=8 name=PCMA channels=1 clockrate=8000/payload-type id=101 name=telephone-event channels=1 clockrate=8000//descriptiontransport xmlns=urn:xmpp:jingle:transports:ice-udp:1//content/jingle/iq - --- XMPP sent to 'google-cathy' --- iq type='result' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='o'/ - --- XMPP sent to 'google-cathy' --- iq from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' type='set' id='j'jingle action='transport-info' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1'content creator='responder' name='audio'transport xmlns='urn:xmpp:jingle:transports:ice-udp:1' pwd='4b4001b575f3c7b824e14d9436d5f466' ufrag='6c28e0a07a5269e82ee313d916a046f7'candidate component='1' foundation='583375015' generation='0' id='0a86' ip='192.168.1.1' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='583378294' generation='0' id='3c7f' ip='192.168.0.10' port='16384' priority='2130706431'
Re: [asterisk-users] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit : Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf Thanks for your reply, unfortunately that makes no difference, I still get: [Nov 5 19:45:16] ERROR[30664][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '14ec70fb484b5700' Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYpNoACgkQuu7Rv+oOo/imrgCgrDUi0VdhCbspzA7SUtFQWpDK iEAAn3X5x/eX96eSRj8PsXqpk4SYFpA5 =98GL -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice and back (chan_motif)
Here are my settings that work. I can make incoming and outgoing calls. Compare my settings with yours. Also make sure your firewall is open for port 5222 and 5060 and your RTP port range. #rtp.conf [general] icesupport=yes rtpstart=15000 rtpend=2 #motif.conf [default](!) disallow=all allow=alaw allow=ulaw allow=h264 transport=google-v1 context=incoming [asterisk](default) connection=asterisk [coopvr](default) connection=coopvr #xmpp.con [asterisk] type=client serverhost=talk.google.com username=coopaster...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Asterisk Server timeout=5 [coopvr] type=client serverhost=talk.google.com username=coo...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Asterisk Server timeout=5 On 11/05/2012 09:49 PM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit : Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf Thanks for your reply, unfortunately that makes no difference, I still get: [Nov 5 19:45:16] ERROR[30664][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '14ec70fb484b5700' Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYpNoACgkQuu7Rv+oOo/imrgCgrDUi0VdhCbspzA7SUtFQWpDK iEAAn3X5x/eX96eSRj8PsXqpk4SYFpA5 =98GL -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roy Abshire Co-op Vacation Rentals 15218 Summit Ave Suite 300-354 Fontana, CA 92336 (855) 760-COOP (4667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice / Jabber auth problem
asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client@theirdom...@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server / configuration will authenticate using a login that is of standard format: username=u...@gmail.com Debugging indicates that the first word in the username field is dropped: jabber set debug on === JABBER: accountone INCOMING: stream:stream from=domain@gmail.com id=C28AAAC2E0 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttlsxmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING: starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: accountone INCOMING: proceed xmlns=urn:ietf:params:xml:ns:xmpp-tls/ JABBER: accountone OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' JABBER: accountone INCOMING: stream:stream from=domain@gmail.com id=3439AAA8B version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client JABBER: accountone INCOMING: stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismPLAIN/mechanismmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'AGNoYXNvbmhvbWVzAGF3Z2MxOTI4/auth JABBER: accountone INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslinvalid-authzid//failure JABBER: accountone INCOMING: /stream:stream JABBER: accountone OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' Is this a bug or can it be made to work somehow? Thank you, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice / Jabber auth problem
Andrew, Did you try username=cli...@theirdomain.tld? -Vladimir On 6/15/2012 9:42 AM, Andrew McRory wrote: asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client@theirdom...@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server / configuration will authenticate using a login that is of standard format: username=u...@gmail.com Debugging indicates that the first word in the username field is dropped: jabber set debug on === JABBER: accountone INCOMING: stream:stream from=domain@gmail.com id=C28AAAC2E0 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttlsxmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING: starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: accountone INCOMING: proceed xmlns=urn:ietf:params:xml:ns:xmpp-tls/ JABBER: accountone OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' JABBER: accountone INCOMING: stream:stream from=domain@gmail.com id=3439AAA8B version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client JABBER: accountone INCOMING: stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismPLAIN/mechanismmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'AGNoYXNvbmhvbWVzAGF3Z2MxOTI4/auth JABBER: accountone INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslinvalid-authzid//failure JABBER: accountone INCOMING: /stream:stream JABBER: accountone OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' Is this a bug or can it be made to work somehow? Thank you, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice / Jabber auth problem
Yes, that and every else I can think of! Thanks. Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 On 6/15/2012 11:31 AM, Vladimir Mikhelson wrote: Andrew, Did you try username=cli...@theirdomain.tld? -Vladimir On 6/15/2012 9:42 AM, Andrew McRory wrote: asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client@theirdom...@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server / configuration will authenticate using a login that is of standard format: username=u...@gmail.com Debugging indicates that the first word in the username field is dropped: jabber set debug on === JABBER: accountone INCOMING:stream:stream from=domain@gmail.com id=C28AAAC2E0 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttlsxmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING:starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: accountone INCOMING:proceed xmlns=urn:ietf:params:xml:ns:xmpp-tls/ JABBER: accountone OUTGOING:?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' JABBER: accountone INCOMING:stream:stream from=domain@gmail.com id=3439AAA8B version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client JABBER: accountone INCOMING:stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismPLAIN/mechanismmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING:auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'AGNoYXNvbmhvbWVzAGF3Z2MxOTI4/auth JABBER: accountone INCOMING:failure xmlns=urn:ietf:params:xml:ns:xmpp-saslinvalid-authzid//failure JABBER: accountone INCOMING:/stream:stream JABBER: accountone OUTGOING:?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' Is this a bug or can it be made to work somehow? Thank you, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice / Jabber auth problem
It looks we have to change the name as two @ appears to break the rules... http://xmpp.org/rfcs/rfc3920.html#addressing Thanks, Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 On 6/15/2012 11:47 AM, Andrew McRory wrote: Yes, that and every else I can think of! Thanks. Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 On 6/15/2012 11:31 AM, Vladimir Mikhelson wrote: Andrew, Did you try username=cli...@theirdomain.tld? -Vladimir On 6/15/2012 9:42 AM, Andrew McRory wrote: asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client@theirdom...@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server / configuration will authenticate using a login that is of standard format: username=u...@gmail.com Debugging indicates that the first word in the username field is dropped: jabber set debug on === JABBER: accountone INCOMING:stream:stream from=domain@gmail.com id=C28AAAC2E0 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttlsxmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING:starttls xmlns='urn:ietf:params:xml:ns:xmpp-tls'/ JABBER: accountone INCOMING:proceed xmlns=urn:ietf:params:xml:ns:xmpp-tls/ JABBER: accountone OUTGOING:?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' JABBER: accountone INCOMING:stream:stream from=domain@gmail.com id=3439AAA8B version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client JABBER: accountone INCOMING:stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismPLAIN/mechanismmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features JABBER: accountone OUTGOING:auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='PLAIN'AGNoYXNvbmhvbWVzAGF3Z2MxOTI4/auth JABBER: accountone INCOMING:failure xmlns=urn:ietf:params:xml:ns:xmpp-saslinvalid-authzid//failure JABBER: accountone INCOMING:/stream:stream JABBER: accountone OUTGOING:?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='domain@gmail.com' version='1.0' Is this a bug or can it be made to work somehow? Thank you, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice STUN error?
FWIW, Thought I searched extensivly with tcpdump and strace, I never found any network traffic that would suggest the error was valid. An upgrade from from 1.8.7.1 to 1.8.10.0 cleared it all up. Thank you, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- Original Message --- From: Andrew McRory andrew.mcr...@sayso.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 1 Mar 2012 14:18:24 -0500 Subject: [asterisk-users] Google Voice STUN error? I have been playing with gvoice over the past few months and it's been great except for this error that appears ONLY when my firewall is enabled: [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #0 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #1 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #2 failed error -1, retry The firewall is configured as documented here http://support.google.com/code/bin/answer.py?hl=enanswer=62464 I've also tried to find the offending packets with tcpdump but have had no luck. Anyone have any bright ideas? Thanks, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice STUN error?
I have been playing with gvoice over the past few months and it's been great except for this error that appears ONLY when my firewall is enabled: [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #0 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #1 failed error -1, retry [Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request: ast_stun_request send #2 failed error -1, retry The firewall is configured as documented here http://support.google.com/code/bin/answer.py?hl=enanswer=62464 I've also tried to find the offending packets with tcpdump but have had no luck. Anyone have any bright ideas? Thanks, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice calling dial plan question.
On Tue, Dec 6, 2011 at 4:05 PM, white hat whitehat...@gmail.com wrote: Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Yes, to both of the last questions. I am using STUN and my asterisk(s) are behind a NAT device (a Netgear WND3700). My jabber.conf looks like: [general] autoregister=yes debug=yes autoprune=no auth_policy=accept [asterisk] type=client serverhost=talk.google.com ; username=xxx...@gmail.com/Talk username=xx...@gmail.com/asterisk secret=XX priority=1 port=5222 usetls=yes usesasl=yes buddy=xxx...@gmail.com status=available statusmessage=I am an Asterisk Server timeout=100 context=gtalk_incoming and, gtalk.conf looks like this: [general] context=LocalSets ; Context to dump call into bindaddr=0.0.0.0; Address to bind to allowguests=yes ; Allow calls from people not in list of peers [guest] ; special account for options on guest account disallow=all allow=ulaw context=gtalk_incoming [XX] username=xxx...@gmail.com disallow=all allow=ulaw context=gtalk_incoming connection=asterisk And, I think that just dumps incoming calls into the context that I posted previously. HTH, dwa -- + dai...@pervasivetelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice calling dial plan question.
On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice calling dial plan question.
If I understand correctly, turning off Call Screening in your Google
Voice configuration should directly connect incoming calls and eliminate
the need to press one.
JF
On 12/2/2011 11:59 PM, white hat wrote:
When a caller calls my google voice phone number, I must answer, wait
and press one to accept. Sometimes even that does not work.
I have tried a few different things to get asterisk to place the call
in an answered state and send the DTMF 1 with the Dial macro.
I found Malcom Davenports wiki page regarding Google calling which has
been very helpful in troubleshooting the issue.
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google?focusedCommentId=18415969#comment-18415969
I'm sure that I'm close to getting things working properly.
Here's my config.
##jabber.conf##
[general]
debug=no
autoprune=no
autoregister=yes
[whitehat238]
type=client
serverhost=talk.google.com http://talk.google.com
username=whitehat...@gmail.com/Talk http://whitehat...@gmail.com/Talk
secret=password
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage=No Information Available
timeout=100
keepalive=yes
##gtalk.conf##
[general]
allowguest=yes
context=googlein
stunaddr=stun01.sipphone.com http://stun01.sipphone.com
[guest]
disallow=all
allow=ulaw
connection=whitehat238
context=googlein
##extensions_custom.conf##
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,1,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,GotoIf($[${CALLERID(name):0:2} !=
+1]?notrim)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,Set(CALLERID(name)=${CALLERID(name):2})
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten = whitehat...@gmail.com mailto:whitehat...@gmail.com,n,Answer
exten = whitehat...@gmail.com mailto:whitehat...@gmail.com,n,Wait(1)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,SendDTMF(1)
exten = whitehat...@gmail.com
mailto:whitehat...@gmail.com,n,Goto(from-trunk,5025551212,1)
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com
mailto:exten...@voice.google.com)
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
I have a working inbound route which rings an internal extension
(7008) when calling the GV number. I can also make outbound calls to
any number using the GV trunk.
I found this page (Link to Michigan telephone blog) which helped me
get everything setup initially and included a shell script that made
it easy to generate the configuration.
http://michigantelephone.wordpress.com/2011/01/20/a-bash-script-to-assist-asterisk-1-8freepbx-2-8-users-in-adding-new-google-voice-accounts/
The author explains the config in more detail and why he choose to
write it the way he did.
I have tried using the alternative method of sending the DTMF 1 tone
by changing the last block as follows:
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com
mailto:exten...@voice.google.com,D(:1))
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)|
However, that did not work.
I just need a little advice on how to write the dial plan. I still
have much to learn about asterisk, and appreciate any advice.
Thanks,
|
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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Re: [asterisk-users] google voice calling dial plan question.
Hey Josh, I've messed with the google voice account settings extensively. As of now, in Google voice account settings I have. Voice tab: forward calls to Google chat checked. Nothing else is checked. Calls tab: call screening is off. On incoming call, display callers number. On Caller ID outing. Don't change anything is selected. Do not disturb is disabled. Nothing else is checked (enabled) The behavior is that the call comes in, and asterisk rings extension 7008, but I never here the prompt by Google to press one to accept the call. It either isn't played, isn't recognized, by Google when asterisk sends the DTMF 1, or it's played before I answer the extension and I don't hear it because the audio streams were not connected when it was played. If I answer extension 7008, and then press 1 (full one second press of the button) then most of the time it will connect the call. Sometimes I have to press 1 two or three times before it will connect, and rarely, it won't connect at all, even with the key presses. As part of the troubleshooting I have removed all other Google voice accounts in extensions_additional.conf, and left only the whitehat238 gvoice connection. Now the prompt is never played but the key press is still required as if it were. On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.comwrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice calling dial plan question.
dwa As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped. I'm sure it's a dial plan configuration issue. Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Thanks On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.comwrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice calling dial plan question.
You could also try putting a Progress() statement between Answer and Wait. I know there is a latency issue with DAHDI calls; 5 seconds may or may not be enough for googlevoice. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of white hat Sent: Tuesday, December 06, 2011 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] google voice calling dial plan question. dwa As part of the troubleshooting I updated all of the asterisk packages from the repo with yum. I'm using freepbx distro (centos based) with asterisk 1.8 There were several newer asterisk 1.8 packages available. I'm not using any custom modules in freepbx. After the updates, I restarted asterisk with core restart now but this hasn't helped. I'm sure it's a dial plan configuration issue. Would you be willing to post sanitized versions of your jabber.conf, gtalk.conf and details regarding the context you're using and how your inbound route is configured in your dial plan? Are you using STUN? Is Asterisk behind a NAT device or on a public IP? Thanks On Tue, Dec 6, 2011 at 10:36 AM, Dave Aibel dai...@pervasivetelecom.com wrote: On Sat, Dec 3, 2011 at 12:59 AM, white hat whitehat...@gmail.com wrote: When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I just need a little advice on how to write the dial plan. I still have much to learn about asterisk, and appreciate any advice. Geez, Maybe I am just brute forcing it, but, the following dialplan seems to work (at least, most of the time!): [gtalk_incoming] exten = s,1,Answer() exten = s,n,Wait(5) exten = s,n,SendDTMF(1) exten = s,n,Dial(SIP/Ciscofficephone,10) exten = s,n,Playback(vm-nobodyavail) exten = s,n,Playback(vm-pls-try-again) same = n,Hangup() HTH, dwa dai...@pervasivetelcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] google voice calling dial plan question.
When a caller calls my google voice phone number, I must answer, wait and
press one to accept. Sometimes even that does not work.
I have tried a few different things to get asterisk to place the call in an
answered state and send the DTMF 1 with the Dial macro.
I found Malcom Davenports wiki page regarding Google calling which has been
very helpful in troubleshooting the issue.
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google?focusedCommentId=18415969#comment-18415969
I'm sure that I'm close to getting things working properly.
Here's my config.
##jabber.conf##
[general]
debug=no
autoprune=no
autoregister=yes
[whitehat238]
type=client
serverhost=talk.google.com
username=whitehat...@gmail.com/Talk
secret=password
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage=No Information Available
timeout=100
keepalive=yes
##gtalk.conf##
[general]
allowguest=yes
context=googlein
stunaddr=stun01.sipphone.com
[guest]
disallow=all
allow=ulaw
connection=whitehat238
context=googlein
##extensions_custom.conf##
exten = whitehat...@gmail.com
,1,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten = whitehat...@gmail.com,n,GotoIf($[${CALLERID(name):0:2} !=
+1]?notrim)
exten = whitehat...@gmail.com,n,Set(CALLERID(name)=${CALLERID(name):2})
exten = whitehat...@gmail.com
,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten = whitehat...@gmail.com,n,Answer
exten = whitehat...@gmail.com,n,Wait(1)
exten = whitehat...@gmail.com,n,SendDTMF(1)
exten = whitehat...@gmail.com,n,Goto(from-trunk,5025551212,1)
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com)
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
I have a working inbound route which rings an internal extension (7008)
when calling the GV number. I can also make outbound calls to any number
using the GV trunk.
I found this page (Link to Michigan telephone blog) which helped me get
everything setup initially and included a shell script that made it easy to
generate the configuration.
http://michigantelephone.wordpress.com/2011/01/20/a-bash-script-to-assist-asterisk-1-8freepbx-2-8-users-in-adding-new-google-voice-accounts/
The author explains the config in more detail and why he choose to write it
the way he did.
I have tried using the alternative method of sending the DTMF 1 tone by
changing the last block as follows:
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com,D(:1))
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
However, that did not work.
I just need a little advice on how to write the dial plan. I still have
much to learn about asterisk, and appreciate any advice.
Thanks,
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Re: [asterisk-users] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 7:58 AM, Tim Panton t...@westhawk.co.uk wrote: On 15 Jun 2011, at 23:29, Kevin P. Fleming wrote: On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- A quick (uneducated) look at the packet, I think google have added some jingle compatibility to gtalk. The packet invite now contains 2 nodes - one in the jingle namespace and one in the google/session namespace this confuses asterisk and it passes the call to _neither_ . I'm not up on iksemel - but I think that if it were told to match on either node, not just the first one things might work again The good news is that it supports a load of nice codecs now, including g722 :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk So I guess incoming calls from gTalk aren't working then? (using v1.8.5.0) I am having the exact same issue as the OP where the outgoing calls work fine but not incoming which never hit any context within Asterisk and the calling party only continues to hear a ringback even thought I can see the jabber debug output for the incoming call on the console. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 3:12 PM, Tim Panton t...@westhawk.co.uk wrote: You should probably not mention the voipusersconfere...@gmail.com address this for week's VUC as at the moment the gateway ignores any calls to it. If/when it comes back to life, we can realistically expect wideband through to zipdx. This said, I see that http://Bluejeans.com/vuc works with Gtalk so we'll see if anyone shows up there today or Tues-Wed. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On 15 Jun 2011, at 23:29, Kevin P. Fleming wrote: On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- A quick (uneducated) look at the packet, I think google have added some jingle compatibility to gtalk. The packet invite now contains 2 nodes - one in the jingle namespace and one in the google/session namespace this confuses asterisk and it passes the call to _neither_ . I'm not up on iksemel - but I think that if it were told to match on either node, not just the first one things might work again The good news is that it supports a load of nice codecs now, including g722 :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton t...@westhawk.co.uk wrote: The good news is that it supports a load of nice codecs now, including g722 :-) And you know what that means? :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On 23 Jun 2011, at 13:44, randulo wrote: On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton t...@westhawk.co.uk wrote: The good news is that it supports a load of nice codecs now, including g722 :-) And you know what that means? Unfortunately it means it doesn't work (yet). You should probably not mention the voipusersconfere...@gmail.com address this for week's VUC as at the moment the gateway ignores any calls to it. If/when it comes back to life, we can realistically expect wideband through to zipdx. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
Hey Elliot;
Would you mind posting your dialplan for your Google Voice config? I am
having a hell of a time getting it to do *anything*.
Perhaps I am just fat-fingering.
Would you mind? Thanks in advance.
Glen
On 6/13/2011 19:02, Elliot Murdock wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING:iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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Re: [asterisk-users] Google Voice receiving call problem
Hello,
Yes, the issue I am having is currently only with Google Talk. Wonder
if what development will be made to fix this issue.
--Elliot
On Wed, Jun 15, 2011 at 9:20 AM, Vladimir Mikhelson v...@mikhelson.com wrote:
Elliot,
I do not think Issue # 17993 is related. As Terry Wilson says on the
Bug Tracker, Google Voice inbound calls still work, it is just coming
from Google Talk that doesn't.
-Vladimir
On 6/14/2011 5:51 PM, Elliot Murdock wrote:
Hello,
Seems that it's been spotted and tracked at
https://issues.asterisk.org/jira/browse/ASTERISK-17993
--Elliot
On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson v...@mikhelson.com
wrote:
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport
fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk
box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are
the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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asterisk-users mailing list
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Re: [asterisk-users] Google Voice receiving call problem
On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
You should probably grab a free DID as a failover from gtalk. Have gvoice ring them both and answer the one that comes through first. In my tests. I have better luck with the DID than with gtalk. -- cobra2 Http://linuxindixie.info Kevin P. Fleming kpflem...@digium.com wrote: On 06/15/2011 04:40 PM, Elliot Murdock wrote: Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue. At some point it will be fixed, and then Google will break it again. Google Talk/Google Voice connections to Asterisk will always be at the mercy of Google changing the protocol, which they do whenever they feel like it and with no warning. In other words, you better not be relying on it for critical communications, and you'll need to be patient when it breaks... because the developers can't just drop everything and fix it when Google changes the protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Google Voice receiving call problem
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
--
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asterisk-users mailing list
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Re: [asterisk-users] Google Voice receiving call problem
Hello,
Seems that it's been spotted and tracked at
https://issues.asterisk.org/jira/browse/ASTERISK-17993
--Elliot
On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson v...@mikhelson.com wrote:
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could be the problem?
Elliot
On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport
fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk
box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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Re: [asterisk-users] Google Voice receiving call problem
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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Re: [asterisk-users] Google Voice receiving call problem
You must have 1.8+ its already been posted the 1.6 didn't get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING: iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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[asterisk-users] Google Voice outbound Caller ID broken
Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
What kind of broken are you seeing. It could be the ID is pseudo ID and may never reflect the actual caller. CF _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Thursday, February 24, 2011 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Google Voice outbound Caller ID broken Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle
Sent: Thursday, February 24, 2011 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Google Voice outbound Caller ID broken
Anybody else noticed that caller id for outbound calls via Google Voice seems
to be broken? It seems to be a Google Voice problem though, not an asterisk
issue.
Yes.. google it
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in
gtalk.conf)/+1(googlevoice#)@voice.google.com,30,D(ww2www${EXTEN:1}#w))
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Re: [asterisk-users] Google Voice outbound Caller ID broken
On Thu, Feb 24, 2011 at 9:08 AM, William Stillwell
will...@stillwellsoft.com wrote:
Yes.. google it
I did. :)
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in gtalk.conf)/+1(googlevoice#)@
voice.google.com,30,D(ww2www${EXTEN:1}#w))
I must have missed that posting. I'll go back and dig it up. Thanks.
--
Chris
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Re: [asterisk-users] Google Voice outbound Caller ID broken
Chris, Can you please provide more details. What do you exactly mean by broken? Do your call recipients get a random CID? Have you tried to call from the GMail WEB interface? Are you getting the same result? -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
Chris, Let me summarize: 1. GV Outbound CID shows Unknown, Unavailable, Out of area (depending on a recipient's carrier) starting some time around 02/15/2011 if a call is placed via Google Chat/Google Talk/Google Mail/Asterisk GTalk channel. See http://www.google.com/support/forum/p/voice/thread?tid=49c21d292e80ff65hl=enstart=40 for other users' accounts. 2. This CID feature failure affects only some GV phones, some still work fine as of 02/24/2011. 3. Calls placed with GV call-back facility work fine for the phones affected by the issue described in #1. 4. A workaround is to set Caller ID (incoming) to Display my Google Voice number As expected it will suppress an incoming CID. So it is not a perfect workaround. 5. Another workaround is to trigger a GV callback facility per William Stillwell's posting. Connection time increases with this workaround. Historically it took 15 days to a month for Google to fix similar problems. -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
Further analysis showed that a call placed using a GTalk channel which came as Restricted was not recorded under History / Placed in Google Voice. A call placed using the same GTalk trunk an hour later was terminated to the same recipient's phone with the proper CID. It looks like a call routing issue on the Google Voice end to me. -Vladimir On 2/24/2011 10:40 PM, Vladimir Mikhelson wrote: Chris, Let me summarize: 1. GV Outbound CID shows Unknown, Unavailable, Out of area (depending on a recipient's carrier) starting some time around 02/15/2011 if a call is placed via Google Chat/Google Talk/Google Mail/Asterisk GTalk channel. See http://www.google.com/support/forum/p/voice/thread?tid=49c21d292e80ff65hl=enstart=40 for other users' accounts. 2. This CID feature failure affects only some GV phones, some still work fine as of 02/24/2011. 3. Calls placed with GV call-back facility work fine for the phones affected by the issue described in #1. 4. A workaround is to set Caller ID (incoming) to Display my Google Voice number As expected it will suppress an incoming CID. So it is not a perfect workaround. 5. Another workaround is to trigger a GV callback facility per William Stillwell's posting. Connection time increases with this workaround. Historically it took 15 days to a month for Google to fix similar problems. -Vladimir On 2/24/2011 8:51 AM, Chris Gentle wrote: Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice + asterisk: calls made to GV# processed but weird
Consider this RESOLVED thanks to the help of [David Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new wiki entry from [Malcolm Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google). The trick was the following in extensions.conf: exten = s,1,Answer() exten = s,n,Wait(2) ;; THIS exten = s,n,SendDTMF(1) ;; AND THIS ARE NEEDED exten = s,n,Background(tnttspWelcome) exten = s,n,Background(CurrentAnnouncement) exten = s,n,Goto(0,1) -- Vinh On Tue, Oct 26, 2010 at 7:07 PM, Vinh Nguyen vinhdi...@gmail.com wrote: Can anyone reproduce this with their google voice number? Wondering whether this issue is just me or not, or whether I am misunderstanding the capabilities of incorporating GV with asterisk. Thanks. Vinh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] google voice + asterisk: calls made to GV# processed but weird
On Thu, Oct 28, 2010 at 7:30 PM, Vinh Nguyen vinhdi...@gmail.com wrote: Consider this RESOLVED thanks to the help of [David Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new wiki entry from [Malcolm Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google). I managed to finally get a GV number while at Astricon. I hope to play with this more next week. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] google voice + asterisk: calls made to GV# processed but weird
Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an auto-attendant on asterisk. I am doing a basic Hello world example. My config: jabber.conf: [general] debug=yes autoprune=no autoregister=yes [asterisk] type=client serverhost=talk.google.com username=mya...@gmail.com/gmail secret=MYPASSWORD port=5222 usetls=yes usesasl=yes statusmessage=Connected to Asterisk. ;required do not change timeout=100 gtalk.conf: [general] context=default allowguest=yes bindaddr=0.0.0.0 [guest] disallow=all allow=ulaw connection=asterisk extensions.conf: [general] [globals] [incoming] exten = s,1,Answer() exten = s,n,Playback(hello-world) exten = s,n,Hangup() [default] include = incoming Basically, when I'm logged into another gmail account and call the computer that's connected to asterisk, the Hello world example works. However, if I call the GV # from a phone, GV rings and end up at the GV voicemail. At first I thought it just skipped the pickup altogether. However, thanks to the help of p3nguin, pabelanger, and [TK]D-Fender on #asterisk, I found out that the call IS processed by asterisk; however, the user does not hear any of it and goes straight to the GV voicemail. I wanted to give the mailing list a try to see if other people have thoughts on this. Here is the debug: [Oct 24 21:18:23] VERBOSE[2393] config.c: == Parsing '/etc/asterisk/logger.conf': [Oct 24 21:18:23] DEBUG[2393] config.c: Parsing /etc/asterisk/logger.conf [Oct 24 21:18:23] VERBOSE[2393] config.c: == Found [Oct 24 21:18:23] VERBOSE[2393] logger.c: Asterisk Queue Logger restarted [Oct 24 21:18:28] VERBOSE[2405] res_jabber.c: JABBER: Keep alive packet [Oct 24 21:18:44] VERBOSE[2405] res_jabber.c: JABBER: asterisk INCOMING: presence from=cal...@gmail.com/androidfe2b05b6ebb0 to=myusern...@gmail.compriority24/prioritycaps:c node=http://www.android.com/gtalk/client/caps; ext=pmuc-v1 ver=1.1 xmlns:caps=http://jabber.org/protocol/caps/status/x xmlns=vcard-temp:x:updatephoto3c4fd5045a18d7417b2e4371bdce077ecd6c8355/photo/x/presence [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: I am available ^_* 13 [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: type is available [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: JABBER: Handling paktype PRESENCE [Oct 24 21:18:44] DEBUG[2405] res_jabber.c: XML parsing successful [Oct 24 21:18:49] VERBOSE[2405] res_jabber.c: JABBER: asterisk INCOMING: iq from=+1caller10di...@voice.google.com/srvres-MTAuMjE4LjIwLjE0Mzo5ODEy to=myusern...@gmail.com/gmail02D370A8 id=jingle:10.218.20.143-28982014:1:C3955FF7 type=setses:session type=initiate id=sip183646...@10.218.118.3 initiator=+1caller10di...@voice.google.com/srvres-MTAuMjE4LjIwLjE0Mzo5ODEy xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq [Oct 24 21:18:49] DEBUG[2405] res_jabber.c: JABBER: Handling paktype IQ [Oct 24 21:18:49] DEBUG[2405] chan_gtalk.c: The client is guest for alloc [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] res_rtp_asterisk.c: Allocated port 11262 for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: RTP instance '0x1b86bc8' is setup and ready to go [Oct 24 21:18:49] DEBUG[2405] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Setting payload 0 based on m type on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Setting payload 101 based on m type on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Incorporating payload 0 on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Incorporating payload 101 on 0x1b86d90 [Oct 24 21:18:49] DEBUG[2399] devicestate.c: No provider found, checking channel drivers for Gtalk - +1CALLER10DIGIT [Oct 24 21:18:49] DEBUG[2399] devicestate.c: Changing state for Gtalk/+1CALLER10DIGIT - state 2 (In use) [Oct 24 21:18:49] DEBUG[2399] devicestate.c: device 'Gtalk/+1CALLER10DIGIT' state '2' [Oct 24 21:18:49] VERBOSE[4341] pbx.c: == Starting Gtalk/+1CALLER10DIGIT-12d0 at default,myusern...@gmail.com,1 failed so falling back to exten 's' [Oct 24 21:18:49] DEBUG[4341] pbx.c: Launching 'Answer' [Oct 24 21:18:49] VERBOSE[4341] pbx.c: -- Executing [...@default:1] Answer(Gtalk/+1CALLER10DIGIT-12d0, ) in new stack [Oct 24 21:18:49] DEBUG[2434] app_queue.c: Device
[asterisk-users] Google Voice-like feature.
I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio wrote: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken You can get them to acknowledge by executing a macro when the call is connected using the M parameter in the dial command. However if the mobile was answered and the confirmation not entered you would have to flag that destination as being dead and then jump back to the dial command again and omit that destination for the next attempt. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
On Thursday 02 Sep 2010, Ken D'Ambrosio wrote: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. The problem is that, if one of the destination phones is diverting to voicemail, you won't know it's voicemail until it's answered -- by which time it's already too late. The best you could hope to do is: park the incoming call; ring all the handsets at once; and when each one answers, play a recorded message giving the number to pick up the parked call. If any of them successfully picks up the parked call, then of course you need to abort the Dial() to the other ones. If no-one picks up the parked call within a reasonable timeframe, it can be sent to Asterisk's own voicemail. -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Subject: [asterisk-users] Google Voice-like feature. I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? This might work - Exten = 1234,1,Dial(DAHDI/1/w#1#2#3,30,p) The Privacy mode switch on the dial would make the called party have to press 1 to accept the call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio ken at jots.org writes: I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Take a look at Followme() and followme.conf. Lonnie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice-like feature.
Want to thank everyone who mailed; a couple of your ideas got me going down certain paths, and found the answer here: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Again, thanks! -Ken original message - I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
