Originally posted by i-p-tel
@acs1122: Your System Log shows registration attempts to acs1122-200 while your account only has extension -100.
I don't understand (sorry). How do I fix it? What do I need to do? Configure Trunk, inbound, outbound...what? Also, what should I submit in the fields?
I have no idea because my account was setup automatically when I installed Sipdroid on my phone. I logged in and it was already working.
The problem began when my account was deleted and I haven't been able to make or receive calls for several days now. I can send and receive text on android phone using GV. But still no calls...
UPDATE: I lost the ability to text now. I paid for personal assistance. Please fix my account
This post has been edited 2 time(s), it was last edited by acs on 11.03.2012 at 01:11.
google voice calling is not working for me. i've tried using csipsimple and sipdroid on an android phone. with both i get a message "your call cannot be completed as dialed, please try again later." the "system log" seems to contain some error messages. it doesn't seem to matter what number i try calling. i've tried using two outbound google trunks (one a gmail account and the other a google apps account). i've tried deleting and re-creating the trunks, routes, and extensions.
Google voice inbound calls to PBXes is not working for me. When calling both GV lines from an outside line I get "The google voice subscriber is not available...." .
One line I have it inbound routed to an extension that forwards all calls to a conference room. I tested dialing the extension and the conference room answers.
The other line I have it inbound routed to an extension. I have eyebeam sip registered to that extension and active.
I call either line from outside and neither answers. I've restarted the pbx twice and it doesn't improve the situation.
I would really appreciate your help as I was trying to show a friend why he should get a premium pbxes account and now I can't
Setting up my GTalk, works fine. How does one setup Inbound Routing so that call forwards directly to Google VM in an "After Hours" time slot. In other words, at night call will not be sent to SIP extension but to GVM?
As of today, it appears that Google Peering is not working. I have tried to re-register my account. I can dial out on my Grandstream device but not on my Google Trunk. Please look into this.
For the white noise, do a "submit and restart" on personal data.
To route to gv voicemail, setup a classic extension that dials the gv # (its like calling your gv # from gmail, or calling your cell number from itself). Then route to it after hours.
Your default outbound must be gv (if not then route calls to your gv # to go out the gv outbound trunk.
For the "undefined default outbound route error", create a blank route (name & rules) that goes to the trunk.
Also I figured out how to be able to both "leave a voicemail" on my google voice account as well as login to my voicemail from the digital receptionist. I create a "login to gv" classical extension that rings "1XXXXXXXXXXpwwww*", where XXXXXXXXXX is my google voice number (it will only work when using the Google Talk trunk you're registered to. This dials it by pressing * after two seconds. To leave a voicemail, simply dial your GV number.
BTW, can I register to two google talk trunks, either inbound or outbound?
This post has been edited 3 time(s), it was last edited by jus on 05.06.2012 at 03:24.
I cannot get this working on my phone anymore, even tho it used to work. When I try to set it up, it says "Account registration failed: incorrect username and password", and I have tried over and over again to confirm i'm typing the correct password, I even changed the password so that I knew it was the same. I am using morrisisaacl-200 as my username, and my Sipdroid extension is 200, why is this not working? When I checked the logs it says "Bad authentication from [email protected]". I am a paid subscriber, but obviously won't be for much longer if I can't get this working
I'm relatively new to PBXES, just purchased a Soho upgrade so that I could hook up google voice.
I'm running CsipSimple on two android devices, when I call out on either device it rings and rings on my end but it does not ring on the receiving end.
In the system log it says:
Jun 21 15:57:03 NOTICE[35345] res_jabber.c: Malformed Jabber ID : ***/Talk (domain missing?)
Did I miss something?
-----
UPDATE:
Figured it out. User error of course. In the Google Voice trunk setup I was entering my user name without adding "@gmail.com" to the end so it wouldn't login. Everything is working perfectly now!
This post has been edited 1 time(s), it was last edited by dec on 22.06.2012 at 21:31.
It's impossible to tell when these forum posts were posted. The only prominent address I can see is the user's registration date. With so much changing in this field, it should be clear. I don't think a posting from 2010 would be 100% relevant; at least a way to sort with latest posting first.
Hello, I recently setup GV with PBXes paid acct and it is working very well I must say (Inbound/Outbound/Auto Attendant etc). I did have the white noise issue but switched to "de" as mentioned here and that was resolved. I do however have one problem...
I am attempting to use the built-in PBX VM for all extensions, but the incoming calls drop immediately and never even get to the VM greeting/message.
On my initial test extension (101), I notice that if my SIP device is not connected, or I ignore the call from Softphone after ring; the same happens and call is just dropped.
Would appreciate any assistance on resolving as I am really excited to finally have a system that just works out of the box like this with GV.
Been enjoying the GV Peering happily for ~2 years now, thx!
However, 1-2w ago it has stopped working. I have 5 extensions (using 4 different GV trunks) and the trunks have all stopped working.
Calls out give the "cannot-complete-as-dialed" error. (system log messages# 20956-20970 demonstrate a failed call.)
Calls to any GV # just ring 20-30 seconds then to VM.
GV credentials are entered as [email protected], passwords have not changed. 3 accounts @gmail, 1 account is a non-gmail, google account.
I re-entered credentials for my GV trunk and log shows the following...
code:
Jan 24 10:59:34 NOTICE[113336] cdr.c: CDR simple logging enabled.
Jan 24 10:59:34 NOTICE[113336] config.c: Registered Config Engine mysql
Jan 24 10:59:34 NOTICE[113336] cdr_addon_mysql.c: MySQL database table not specified. Assuming "cdr"
Jan 24 10:59:39 NOTICE[74051] cdr.c: CDR simple logging enabled.
Jan 24 10:59:39 NOTICE[74051] config.c: Registered Config Engine mysql
Jan 24 10:59:39 NOTICE[74051] cdr_addon_mysql.c: MySQL database table not specified. Assuming "cdr"
Jan 24 10:59:44 NOTICE[74057] res_jabber.c: JABBER: encryption failure. possible bad password.
Forum search for "encryption failure" only returned this thread, but I couldn't find which page mentioned it!
I have also 3 accounts GV attached to PBXes Soho as trunks. No one is working for last 1-2 months.
Calls out give the "You call cannot be complete-as dialed" error.
SysLog NOTICE[114651] app_dial.c: Unable to create channel of type 'GTALK' (cause 0 - Unknown)
What is wrong? Help me please!
In case of authentication errors (the bad password message) with multiple GV trunks, I suggest deleting them, and re-starting with a single GV trunk. Then you are able to add the trunks step-by-step to find out which one gives the bad password message.
Unfortunately the message does not include any hint which GV trunk has the bad password.
Not sure if this is the correct thread. Got issues with the loss audio for google trunks. Incoming calling party tells me that they hear me but unfortunately I can't hear them and so I just hang up. They tried calling back but the same issues exists.
I already tried submit and restart my personal data and removed the audio bypass for each extension but still the problem exists. Sometimes i'm being cut off in the middle of conversation but the call remains active. When I call back telling that I was suddenly cut off, the caller says that the line is still active when I hang up on them. But in reality, I can no longer hear them so I hang up and call them back.
How can I improve or avoid audio loss passing through from google trunk to each sip and classic extensions?
RE: Google Voice Peering
RE: Google Voice Peering
RE: Google Voice Peering