Thread: Google Voice as SIP Provider |
mis
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Hell-O again
After the last update (where you asked us to delete the gvoice trunk, then 'submit & start', etc.), Everything was working smoothly, I can also tell the you either upgraded some systems or got the load issues worked out, too.
Only issue I am having is that bug where you only get 4 rings on outgoing calls before you hear a busy signal and then a hang up...
That problem had gone away for a while, actually... but just popped up again. The only recent change I made was changing the password on my main SIP extension, which I doubt is related.
So it's back to calling people, if they don't answer at 4 rings (when the system seems to go busy and then hang up), I have to text people to let them know why I called, lol.
I tried removing, readding the trunk a couple of times, and also tried hitting 'submit & start' again just for kicks... nothing seems to help.
If anyone has solved this, or if anyone else is still experiencing this glitch, do advise ![Augenzwinkern](images/en/smilies/wink.gif)
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Thread: Google Voice as SIP Provider |
mis
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I am getting really sick of seeing "timed out" when I go to make calls...
Just like the user above, about 1 out of every 4 calls actually goes through.
I have to resave the whole trunk and it gets better for a short while before going back to useless.
It's really irritating on my main business line, which used to work much better before the GVSIP change. I lose at least a customer every day!!!!!!!!!!
This was working for a few days, as I had reported.
Whatever changed these last 4 days, the service has gone downhill...
I know its hard to make changes and roll out a feature so quickly.
However a lot of other providers have this working now.
Tried to help your user base, I know its beta...
But there is little response here. Too little too late.
Seems pretty clear your servers are not handling the requests very well.
I want a refund for the past month subscription.
Oh, and try the damn codec change I suggested earlier and in the Google voice peering thread.
It will make your servers 60% more CPU efficient.
That advice is only free after you refund my subscriptions.
You can try again next month to provide more than 20% uptime.
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Thread: Google Voice as SIP Provider |
mis
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***SEEMS TO BE FIXED, READ WHOLE POST***
Hi there everyone,
After I had it pointed out to me that the GVsip options are integrated into the regular 'add sip trunk' page (sorry about that ), I was able to get everything connected to my GV account.
The only problem is that I am experiencing the same issue as others here... my GV trunk only stays registered for random but short periods of time. Sometimes I manage to get a few test calls in and other times it dies so quickly I don't get a chance.
I checked the system log and it doesn't reveal much other than that the GV account is no longer registered (which I already knew).
NOTICE[44694] chan_sip.c: -- Registration for 'gvuser@obihai.sip.google.com' timed out, trying again (Attempt #16)
VERBOSE[44694] logger.c: -- Got SIP response 603 "Rejected"
***read my edit!
The random time interval before the trunk becomes unregistered makes me wonder if it is a bottleneck of some kind on the server side.
While experimenting with naf's (and friends) pjsip implementation, I noticed that I had to use the 'opus' codec on the extension side of the connection to prevent my test system from bogging due to transcoding load. I was using a small VM, though...
Then again, it might also just be a timeout bug somewhere...
I am going to see if I can find anything else about this on the DSL reports page or commit reports on naf's pjsip repository.
If anyone else has ideas, please chime in ![Augenzwinkern](images/en/smilies/wink.gif)
EDIT:
Right after I posted this, the GV trunk magically registered again and stayed that way for about ~1min. After that, it greyed out again. But, I was able to get some relevant lines from the syslog as it stopped working:
VERBOSE[44694] logger.c: Capabilities: us - 0x61e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
VERBOSE[14880] logger.c: -- SIP/XXXMYACCOUNTXXX-acfe answered SIP/myaccount-X00-f084
VERBOSE[14678] logger.c: -- Called XXXMYACCOUNTXXX/18002255288
VERBOSE[44694] chan_sip.c: SIP response 100 to standard invite
VERBOSE[44694] chan_sip.c: SIP response 183 to standard invite
followed by the normal 'hang-up' blah blah blah BUT:
NOTICE[44694] chan_sip.c: -- Registration for 'gvuser@obihai.sip.google.com' timed out, trying again (Attempt #1)
Hey, wait a minute. Isn't ' gvuser ' supposed to be a UNIQUE UUID for each account !?!?! I looked inside of Legacy Google Voice's settings on my google account page and the check box for 'sip:XXXXXXXXXXXXXXXXXXXXXXXX@obihai.sip.google.com definitely is unique and not ' gvuser '. I copied the unique identifier (the XXXXX part above) into the 'username' field in my SIP trunk config and everything seems to work great now. Apparently, the 'get google voice password' button does EXACTLY what it says; it gets you the password and nothing else. You will need to visit the Legacy Google Voice page and copy your username into your SIP trunk settings and save it again... It is odd that Google's servers will let you register a trunk without a valid username, even for such short intervals. This, I am sure, is a vulnerability on Google's side. It would allow anonymous calls and 'ghost' trunks, not to mention possible account impersonations... hope it gets fixed.
I also noticed that the registration service polled the codec support list and it did not have 'opus' in the list... don't know if this is relevant, but opus definitely is lighter on the server.
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Thread: Google Voice as SIP Provider |
mis
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99 |
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169312 |
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That is for removing the broken trunks from my account ![Augenzwinkern](images/en/smilies/wink.gif)
The 'add gvoice trunk' button still seems to be missing ![unglücklich](images/en/smilies/frown.gif)
I just have the two to add a sip or enum trunk.
Is the www2 server missing the functionionality still?
I paid for an additional subscription to the low bandwidth vocodoer and tried to switch From soho to a pro account and still don't see it (and now double payed for my account at this point).
Let me know if there is any trick to make it show up...
Thanks.
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Thread: Google Voice as SIP Provider |
mis
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Hello,
I am really excited to hear that there is so much interest and progress related to the new google voice SIP interop.
I checked my account for the aforementioned 'create trunk' option, but I only saw the buttons for 'SIP trunk' and 'Enum trunk'. The Gtalk trunk is gone now, but there was no 'Google Voice trunk' button at all.
I instead tried to create the trunk by using a URL workaround;
I entered 'https://www2.pbxes.com/config.php?display=6&tech=GVOICE' into my address bar.
This brought up the page to create a google voice trunk, I filled in the necessary info and used the 'get password for google voice' button and that part worked fine. I clicked 'save' then saw the red 'apply changes' bar, and also clicked that. After the settings saved I then got 'DB Error: no such table' and no settings page loaded.
Is this because I have used the URL instead of asking about the missing button first?
In any case, my trunks menu is pretty messed up now...
there are multiple google voice trunk entries that just bring up the 'DB Error: no such table' when I try to click on them. It is also messing up the inbound calling that was still functional on the old google talk trunk.
Any suggestions, or maybe you awesome i-p-tel folks can delete those errant gvoice trunks since I cannot? let me know.
Thanks for all your hard efforts, I am impressed with how much you have implemented on this so quickly...
P.S. I am at the www2 datacenter...
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Thread: RE: Google Voice Peering |
mis
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I seem to be still getting incoming calls okay, when I place a call to my GV number, my SIP extension rings and I can answer the call and converse normally.
If I place an outgoing call from my SIP extension, the other end rings and can accept the call... However it is one-way, the other end can hear me, but I cannot hear them. I have tried dozens of tests on different SIP phones (hardware and software) and re-checked and re-saved my PBXes configurations several times to no avail.
I cannot determine if this is a coincidental bug that happens to time with Google's end of XMPP interop, or if this issue is related to the changes on Google's end.
I would love to keep using this system, as it has worked out pretty well for me to date...
Perhaps I am overlooking something simple and someone can point it out for me.
P.S. I am available and interested in beta testing new interop plugins for Asterisk if I can offer any of my expertise to that effort here...
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Thread: RE: Google Voice Peering |
mis
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Hi everybody!
It is my understanding that as of June18, Google will be cutting the strings on their XMPP interop service. Being that PBXes utilizes the XMPP GV Asterisk plugin to trunk to Google Voice accounts, it seems like this feature will stop working...
As of Yesterday (June 29th), I can no longer make outgoing calls via my Google Voice trunk. I simply get a single ring and then 'dead air'. That is, I hear a ring and then the call just sits open, but does not ring the destination phone or give me any ringbacks or anything.
I can, however, still call my GV number and my SIP extension rings and connects the call.
I need to know if this is because of an issue with my PBXes trunk or extension, or if this is just a symptom of Google shutting down their XMPP interop systems.
I am posting here because I think your user base needs to know what the bottom line is, and if you have any plans to add the new Asterisk plugin that Simon and all of us other users have been beta testing with other Asterisk users (mentioned in a number of forums, not sure if it works yet tho).
Please let me (all of us) know what to expect. If this is just a bug with my trunk, and XMPP is still up, we need to know that, too.
Thanks again, it's been a good run!
P.S. The forum seems to be denying me from posting a new reply, so I will add it here; We have come up with some scripts and plugins to allow any Asterisk based system to trunk the new GVSIP protocol. Everybody here might benefit from this discussion and collection of Asterisk code:
http://nerdvittles.com/?p=26204
http://www.dslreports.com/forum/r3201698...ussion~start=90
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Thread: RE: You Deleted My Gtalk Trunk! |
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I tried a new premium subscription, it definitely does not have the GTALK or GVOICE trunk button. Seems like only SOHO does.
Also, you say GTalk is support intensive. I mentioned (this is the second post) a support issue with my GTalk trunk hanging intermittently. You asked me to check the Google account and make setting changes. I did. It still hangs. It says connected but wont make or receive calls. I have to resave the password twice before it works again.
Looks like I wasted the 16.00 on the subscription which I had to cancel 10 minutes later.
If you want to keep it instead of refunding me, then please just fix the GTalk trunk.
I am willing to pay you instead of just deleting and recreating my account through sipdroid. I have the promo version still in my backups. I just want the GTalk trunk to work.
It feels rough to get the GTalk trunk for free and then it gets deleted. You might work on another business strategy for that so people don't feel rubbed. There is a lot of talk on various forums about it and it's getting you a bad rap.
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Thread: RE: You Deleted My Gtalk Trunk! |
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I have been using pbxes since 2010 and have had a GTalk trunk the entire time. For some spans of time, when I can afford it, I have paid for a premium account to use the g.729 vocoder to save bandwidth.
Recently, my credit card number changed, and I lost my premium subscription.
Several days later, I received a message saying that premium features have been removed from my account.
I logged in and noticed that my GTalk trunk has been deleted.
I got the GTalk trunk originally from your Sipdroid promotion and have been using it ever since.
I am pretty upset that you just randomly decided to delete it after all this time.
So, despite feeling cheated as a customer, I went and re-subscribed to a premium account.
Much to my surprise, I STILL cannot restore my GTalk trunk!
Not to mention, I have had issues with the trunk failing intermittently (which I previously posted about). Every other person I have set up a pbxes.com account for had those issues as well and stopped using the service.
I chose to stick it out, ask for support and just live with it, and still gave you my 16.00 a month.
To make it clear, the ONLY reason I use your service is because of the Google Voice trunking.
Why you would delete the trunk now, after I have had it all these years is beyond my comprehension why it would suddenly get deleted...
Am I being punished for asking for support?
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Thread: RE: GTalk trunk failing intermittently |
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Thanks for taking a look, I really appreciate it.
Another but of information I noticed after monitoring the issue:
It appears that the failure only occurs after I get a call that I do not answer. IE: somebody calls my gtalk line, it rings on my device, I do not answer and let it go to voicemail, and then after that the trunk fails and I am unable to receive or make any subsequent calls after that point.
Maybe this will help with the troubleshooting. Let me know if there are any other developments...
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Thread: RE: GTalk trunk failing intermittently |
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Any chance you've gotten a chance to check out my logs and account settings? Still trying to sort out why I have to resave my gtalk trunk all the time...
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Thread: RE: GTalk trunk failing intermittently |
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I have been using pbxes since 2011 and have recommended the service to a number of colleagues, and am generally pretty happy about it. However, I have had a number of issues over the years with my GTalk trunk randomly disconnecting...
I have a Google Voice account, and a GTalk trunk that connects to it. I use a YeaLink SIP phone to connect to my PBXes extension. Randomly (sometime twice a day, sometimes not for a week or more), calls will fail to connect and I will get a "no response, timed out" error on the phone itself. After this occurs, I will continue to get the "timed out" messages until I "fix" the issue.
To temporarily resolve the issue, I have to log into my PBXes account, click on the offending GTalk trunk, and re-enter and savey password TWICE. After this, calls will connect until the issue randomly presents itself again. I feel it's important to note that when I log in to my PBXes account to "fix" the issue, the GTalk trunk always says "connected" in green, even when it isn't working. At no time does the trunk actually say "disconnected". Over the years I have had several ISPs and different network environments and the issue persists. Even using a softphone on a cell over data connection does not resolve it...
I am using a paid account, although I occasionally switch to free service when I am behind financially. The issue exists in free, premium, and SOHO subscription modes.
Am I missing some important config setting here or what? It is, after years, finally becoming too irritating to have to sign into my account all the time to resave the trunk and I want it properly fixed.
Please advise, thank you.
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