Re: [asterisk-users] Reliable SIP Trunk Provider
I had many of the same problems with sip station. If you just need sip termination, Check out flow route. The service just seems to work properly for me, and they respond to tickets. You can open up new cases through their site. On Mar 15, 2012 11:48 AM, Jake Wicke j...@nxtphase.net wrote: I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk registering too often. Support either does not respond to e-mails, hangs up on phone calls, or gives me the we don't support Asterisk and we can use your account no problem using the SIP phone on our desk line. Coredial resigned me into a two year agreement after making a change to my SIP trunk configuration without my knowledge, then demanded two years of the full monthly charge when I tried to cancel over a dispute regarding services that I did not order. Check out coredialhorrorstory.com for the whole story. While the service is decent, the customer service leaves much to be desired. Broadvox has been the best provider that I have found so far, however I initially had a lot of issues with sales quoting a product which could not be provisioned and also not being able to deliver service on a timely schedule. I also was given the run around by customer service recently on a simple request to add a DID number to an account. Thanks for your input! -- _ -- 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] Client - registers but unreachable
You could also set qualify to no for this extension, or increase the qualify timeout. Does the bb have a reliable signal, and consistent connectivity to the network? When you say the phone is on the same wifi network, do you mean that it's actually the same subnet? Or is the asterisk server on a different internal network. For example, maybe your wireless devices are in a DMZ net and the asterisk server is on another network? If so, you may still need to enable NAT for the extension. You should be able to confirm whether or not this is a NAT problem with tshark or tcpdump on the asterisk server. It will be clear what IP the asterisk server thinks it's talking to in the packet trace. On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote: I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is unreachable. Any suggestions? Is this just a dumb client or do I need to tweak an asterisk setting? Thanks pbx*CLI sip debug peer 230bb Unable to get IP address of peer '230bb' The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. pbx*CLI -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 pbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 230bb/bob 172.31.254.53D 9653 UNREACHABLE -- _ -- 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] Client - registers but unreachable
From looking at the output above, I notice that there may be two subnets in use depending on what the subnet mask is. The output provided shows two possible networks: 172.31.253.0/24 and 172.31.254.0/24. Or is this all part of the same address space with a different mask? If it is all the same space, then is the asterisk server network stack properly configured with a proper subnet mask? The bb can reach the asterisk server because it registers. Hope this helps On Sun, Jan 1, 2012 at 7:30 PM, white hat whitehat...@gmail.com wrote: You could also set qualify to no for this extension, or increase the qualify timeout. Does the bb have a reliable signal, and consistent connectivity to the network? When you say the phone is on the same wifi network, do you mean that it's actually the same subnet? Or is the asterisk server on a different internal network. For example, maybe your wireless devices are in a DMZ net and the asterisk server is on another network? If so, you may still need to enable NAT for the extension. You should be able to confirm whether or not this is a NAT problem with tshark or tcpdump on the asterisk server. It will be clear what IP the asterisk server thinks it's talking to in the packet trace. On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote: I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is unreachable. Any suggestions? Is this just a dumb client or do I need to tweak an asterisk setting? Thanks pbx*CLI sip debug peer 230bb Unable to get IP address of peer '230bb' The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. pbx*CLI -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 pbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 230bb/bob 172.31.254.53D 9653 UNREACHABLE -- _ -- 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.
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
[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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