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] No voice when the calls come from Internet
Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ? Hi Neo, In the documentation look for this options: externip localnet nat It helps to understand them because this kind of situation is common with VOIP. Best. -- _ -- 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] No voice when the calls come from Internet
Hi, I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don't hear any thing! I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP range in rtp.conf forwarded to my Asterisk server. Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ? Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call. Regards, -- _ -- 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] HD Voice -- connecting Asterisk into HD Voice compatible mobile phone
I've just noticed that mobile phones are starting to support HD Voice. http://thenextweb.com/apple/2012/09/21/iphone-5s-hd-voice-impressive-theres-still-work-done/Is there any hardware capable of connecting HD voice quality calls into Asterisk now? I see that Asterisk compatible GSM hardware is available, but does anything support HD voice presently or should in the future? -- _ -- 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 !
--
__**__**_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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--
__**__**_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--
_
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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
Re: [asterisk-users] Incompatible voice frame ulaw/alaw
Hi Larry, Am 26.08.2012 03:57, schrieb Larry Moore: On 26/08/2012 3:45 AM, Markus wrote: When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk? It doesn't seem to have any effect on the voice quality but the messages on the console are quite annoying. I suspect you will find the frequency of these messages is the value you have set for rtpkeepalive. I would suggest you include the following in your peer's configuration; rtpkeepalive=0 thanks! Unfortunately, that didn't change anything. Any other advise? I don't get where it's configured that Asterisk' default hard-coded codec is ulaw... at least that's the information we get from that message, right? Regards Markus -- _ -- 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] Incompatible voice frame ulaw/alaw
Hi list, I'm resending this.. no one answered. No one got an idea? Thank you! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk? It doesn't seem to have any effect on the voice quality but the messages on the console are quite annoying. PS: The peer doesn't support ulaw. PPS: Asterisk 10.7.0 Thanks a lot! Markus -- _ -- 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] Incompatible voice frame ulaw/alaw
On 26/08/2012 3:45 AM, Markus wrote: Hi list, I'm resending this.. no one answered. No one got an idea? Thank you! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk? It doesn't seem to have any effect on the voice quality but the messages on the console are quite annoying. I suspect you will find the frequency of these messages is the value you have set for rtpkeepalive. I would suggest you include the following in your peer's configuration; rtpkeepalive=0 Larry. -- _ -- 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] Incompatible voice frame ulaw/alaw
Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as the native format, adjustable in some config file or is it hard-coded into Asterisk? It doesn't seem to have any effect on the voice quality but the messages on the console are quite annoying. PS: The peer doesn't support ulaw. PPS: Asterisk 10.7.0 Thanks a lot! Markus -- _ -- 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
[asterisk-users] Increasing voice volume without getting echo or entered digit problem
Dears; How I can increase the voice volume in the analogue line (at dahdi port) without getting a problems in the entered digits (when using Background function)? I got to know previously that it is possible to increase the gain at hardware and not software, this is better to avoid getting a problems. But where? I am using DAHDI 2.4 and another machine has DAHDI 2.6 Using txgain and rxgain from chan_dahdi.conf will not help, it will increase the voice volume but with the following problems: 1) Suddenly the call will be disconnected while we are talking. 2) When calling the Asterisk box and we entered the digits, it is failing to collect it (sometime does not collect it correctly and sometime it collects the digit duplicated). 3) Echo problem. So I need to know how to increase the voice volume from another place? Appreciate the kindly help. Regards Bilal -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence. If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer From: bilal ghayyad bilmar...@yahoo.com To: asterisk-users@lists.digium.com Sent: Thursday, May 10, 2012 2:11 PM Subject: [asterisk-users] Increasing voice volume without getting echo or entered digit problem Dears; How I can increase the voice volume in the analogue line (at dahdi port) without getting a problems in the entered digits (when using Background function)? I got to know previously that it is possible to increase the gain at hardware and not software, this is better to avoid getting a problems. But where? I am using DAHDI 2.4 and another machine has DAHDI 2.6 Using txgain and rxgain from chan_dahdi.conf will not help, it will increase the voice volume but with the following problems: 1) Suddenly the call will be disconnected while we are talking. 2) When calling the Asterisk box and we entered the digits, it is failing to collect it (sometime does not collect it correctly and sometime it collects the digit duplicated). 3) Echo problem. So I need to know how to increase the voice volume from another place? Appreciate the kindly help. Regards Bilal -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
/etc/asterisk/chan_dahdi.conf is where you control txgain and rxgain in DAHDI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, May 10, 2012 1:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Increasing voice volume without getting echo or entered digit problem Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
On Thu, May 10, 2012 at 11:36:51AM -0700, bilal ghayyad wrote: When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Yes, that is the correct file. Any *.conf file in /etc/modprobe.d will suffice but dahdi.conf is the convention. You can also accomplish this from the Asterisk CLI with dahdi set hwgain. Type dahdi set hwgain on the Asterisk CLI for more information. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel 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] Increasing voice volume without getting echo or entered digit problem
I am sure there should be another place .. if I increased it from chan_dahdi.conf, the voice quality is bad and the calls will disconnecting while we are talking .. Increasing voice volume from chan_dahdi means increasing it at software level, I am sure there is a place to increase it at hardware level. Let us agree on something: Is settings to increase it at hardware level? In Zapata, it was existed and can be done as mentioned in my previous emails (from modprobe.conf), can we agree on this? If yes, so why it is not possible in dahdi? Regards Bilal /etc/asterisk/chan_dahdi.conf is where you control txgain and rxgain in DAHDI. -Original Message- Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 22:32, Johan Wilfer skrev: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! I contacted them and will use a server connected to a switch-port in mirroring mode. The gui seems like a great tool in troubleshooting. Nobody uses the rtcp-stats in asterisk for quality monitoring? Other suggestions? -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
On Wed, Apr 11, 2012 at 4:29 AM, Johan Wilfer li...@jttech.se wrote: Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Cheap dedicated server with a span port on the switch. We *never* run anything other than Asterisk on a production voice server. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating different options. Aside from the manual thing I could see two variants: - Dump the traffic (on the server or another via switch port mirroring/monitoring) and analyze it with tshark - Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking for some ideas to setup this so I can eliminate this manual and time-consuming process in the future. And know about the problems before the customer complains about the quality.. Thanks in advance! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
Le 09/04/2012 13:42, Johan Wilfer a écrit : After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating different options. Aside from the manual thing I could see two variants: - Dump the traffic (on the server or another via switch port mirroring/monitoring) and analyze it with tshark - Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking for some ideas to setup this so I can eliminate this manual and time-consuming process in the future. And know about the problems before the customer complains about the quality.. At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. To monitor the traffic, you can use voipmonitor.org -- Daniel -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.netwrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. To monitor the traffic, you can use voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] Monitoring voice-quality with sip/rtp/rtcp
OpenVZ is not really virtualisation, though for some reason people insist on throwing it into the same discursive space as Xen, VMware, HyperV, etc. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Johan Wilfer li...@jttech.se wrote: 2012-04-09 20:22, Carlos Alvarez skrev: On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: At first, if your Asterisk is in a VM install it on the real server, it solved us on some installations. We've gone away from VMs altogether. I use openVZ to run multiple asterisks on the same server. This works well and has done for some time. But currently once a week for about 10-15 minutes calls sound like packetloss/jitter occurs. But a week of traffic captures is heavy... So I need to automate this. To monitor the traffic, you can use voipmonitor.org http://voipmonitor.org We purchased the commercial version with a GUI and will tell you that the cost/benefit is very clear. Great tool, pretty cheap ($1k I think). Responsive support. Sounds very reasonable. Do you run this on a dedicated server, and configured the switch to duplicate the traffic to the quality server? Or do you run this on the same server as asterisk? Thanks for the suggestions! -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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] German voice recognition
Hi, I am looking (for the best) solution to recognize *german* words or simple phrases with a given number of words (eins, zwei drei etc. or hauptmenü, zurück etc.). Can somebody give me a good link? Can I find external service providers who can be accessed via ASR()? Best regards, -Thorsten- -- _ -- 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] German voice recognition
To the best of my knowledge, your best options, not necessarily in order are: 1. Vestec ASR 2. Lumenvox ASR 3. google ASR (there was a good post in February about how to use this) 4. Sphynx ASR Options 1 and 2 are/were recommended by Digium. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Monday, March 12, 2012 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] German voice recognition Hi, I am looking (for the best) solution to recognize *german* words or simple phrases with a given number of words (eins, zwei drei etc. or hauptmenü, zurück etc.). Can somebody give me a good link? Can I find external service providers who can be accessed via ASR()? Best regards, -Thorsten- -- _ -- 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
[asterisk-users] Help_video voice mail not retriev properly
Hi Friend I first store voicemail with video using Xlite. Xlite has problem that for storing video ,required to click on start buttom.And when i retreive it, then i find that , video and voice mail have no syncronization.As Example--I record voice 5 sec and after 5 second then click start buttom to record video, thus video and voice both go to record . When i go through (exten = 704,1,VoiceMailMain() ) it play video and audio both ,not play video after 5 second, thus there is asyncronization occure. After that i used mercuro soft phone.It has feature that ,there is no need to click on start buttom to start video recording.thus video and voice record properly.But when i retreive it through VoiceMailMain() , then video play very fast and finish first in comparison to audio. exten = 102,1,VoiceMail( 102@default,u ) exten = 102,n,Hangup() i call to 102 to deposite video mail. Plz tell me ,how i remove this problem. And which softphone I will use ,which is suitable for record video and syncronization can achive. ThanksRegards Durgesh Mishra Mob No +91-9958823712-- _ -- 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,
|
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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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[asterisk-users] No Voice path during NCS call with Asterisk 10.0.0
Hi, I tried to establish NCS call b/w 2 endpoints of same PacketCable MTA using asterisk-10.0.0. I observed that MDCX sent to aaln/1 contains its own SDP. Some I observed with aaln/2. So voice path is not established b/w aaln/1 and aaln/2. My Configurations: mgcp.cong: [mta84.globaledgesoft.com] host= mta84.globaledgesoft.com wcardep = aaln/* callwaiting = 1 ;canreinvite = 1 dtmfmode= rfc2833 ;amaflags= BILLING ncs = yes ; Use NCS 1.0 signalling ;pktcgatealloc = yes ; Allocate DQOS gate on CMTS ;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate callerid= 3341 ;accountcode = test-362265 line= aaln/1 callerid= 3342 ;accountcode = test-362266 line= aaln/2 extension.conf: exten = 3341,1,Dial(MGCP/aaln/1...@mta84.globaledgesoft.com) exten = 3342,1,Dial(MGCP/aaln/2...@mta84.globaledgesoft.com) can anybody help me to resolve this issue. Regards Vikas -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Greetings- I'm working on a unique Asterisk installation where I've been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let's assume I have remote users connecting to an Asterisk server via sometimes unreliable connectivity such as satellite, wireless, or shudder dial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again? In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)? Thanks! --Tim -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Assuming that you don't have some sort of reconnect protocol going on like SIP headers, a native-bridge to a local channel might do the trick for you. If you are using DAHDI, you might be out of luck. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Monday, October 03, 2011 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Keeping Voice Call Active During Data Connectivity Loss Greetings- I'm working on a unique Asterisk installation where I've been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let's assume I have remote users connecting to an Asterisk server via sometimes unreliable connectivity such as satellite, wireless, or shudder dial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again? In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)? Thanks! --Tim -- _ -- 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] Keeping Voice Call Active During Data Connectivity Loss
Maybe you could use a very simple sollution like a meetme room - you have only to be creative with the dialplan. Ioan www.modulo.ro -- _ -- 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 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
[asterisk-users] Festival VOICE MAIL TO FEMAIL
Hi all How can I change the festival application voice in asterisk from mailvoice to femailvoice. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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 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] Goggle voice incoming dialplan
On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.comwrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts ( wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Goggle voice incoming dialplan
Could you elaborate on how you can associate those non-gmail accounts with gchat account? On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.comwrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts ( wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Goggle voice incoming dialplan
I have a free google apps account (http://www.google.com/a I think) setup for SelbyTech.com. Basically it is a gmail account, just with a different domain. Thanks, --Warren Selby, dCAP On Jun 17, 2011, at 2:43 AM, asterisk asterisk aster...@ck-lee.com wrote: Could you elaborate on how you can associate those non-gmail accounts with gchat account? On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.com wrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts (wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.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 -- _ -- 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
