yashsingh1/xdaforums huggingface dataset:
[Nexus 4 General] ttabbal:
I'm referring to GV in general. It's not a ton, and I don't take the time to wireshark or use some other tool to watch the packets. I haven't seen any go as long as several seconds though. It's always been <1second. I notice it without VOIP involved at all, forwarding the call to a direct cell number. Technically, GV is VOIP, but that's out of my ability to influence. I'm referring only to the bits I can change. :)

Using the PIAF EC2 server, testing only on wifi with a nice strong signal, I do get the same thing. Using a dialing code to force voip.ms to handle the call, I don't get it in the same setup. It's a minor issue for me, but something I noticed playing with GV. If you are getting more than slight bumps, it's likely not what I'm referring to. Some people are more sensitive to that sort of thing than I am, that's why I mentioned it.

For general use, particularly over cell data networks, packet loss is pretty common. That's one reason I tried the silk codec. It's got some nice features for dealing with that. For example, near my office the VOIP link is pretty worthless. There's just not a reliable enough transport for the data. Sometimes it works fine, sometimes it doesn't. So I just don't use it there. Other areas, it's flawless. Hopefully Tmobile will light up my area for LTE soon and update the older towers like the one near my office.
[Nexus 4 General] chdloc:
Well, as my particular experiment above shows, major damage is not done on my end but either between my EC2 instance and Google or within the Google servers themselves. Not sure I can narrow that down further other than getting a commercial SIP trunk, i.e. voip.ms as you suggested, and do some side-by-side comparisons.

Oh, and I'm getting paid for being sensitive to audio problems...;)
[Nexus 4 General] acegolfer:
At dslreport, many ppl claimed that GV is not 100% reliable. I would not use GV for business. In fact, if I use my cellphone for business, I would get an unlimited calling plan not the $30 plan.
[Nexus 4 General] bluespire:
If the business is paying for it and you are an employee, sure. If you are a one-man start up, and not making much of anything yet, not so sure. A high-end phone line will cost over $1000 a year.
[Nexus 4 General] easyxtarget:
I can't get this to work for me. Every time I try to make a call I get a message that the call can't be completed as dialed.

Also I realize you aren't supposed to dial a 1 before the number but I get the same output when I don't. That's not the problem.

[2013-03-02 05:46:22] VERBOSE[3870] chan_sip.c: -- Registered SIP '2562' at 71.229.20.210:42109
[2013-03-02 05:46:23] NOTICE[3870] chan_sip.c: Peer '2562' is now Reachable. (136ms / 2000ms)
[2013-03-02 05:46:28] VERBOSE[3435] netsock2.c: == Using SIP RTP TOS bits 184
[2013-03-02 05:46:28] VERBOSE[3435] netsock2.c: == Using SIP RTP CoS mark 5
[2013-03-02 05:46:28] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:1] ResetCDR("SIP/2562-00000007", "") in new stack
[2013-03-02 05:46:28] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:2] NoCDR("SIP/2562-00000007", "") in new stack
[2013-03-02 05:46:28] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:3] Progress("SIP/2562-00000007", "") in new stack
[2013-03-02 05:46:28] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:4] Wait("SIP/2562-00000007", "1") in new stack
[2013-03-02 05:46:29] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:5] Progress("SIP/2562-00000007", "") in new stack
[2013-03-02 05:46:29] VERBOSE[3884] pbx.c: -- Executing [18506446200@from-internal:6] Playback("SIP/2562-00000007", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2013-03-02 05:46:29] VERBOSE[3884] file.c: -- <SIP/2562-00000007> Playing 'silence/1.gsm' (language 'en')
[2013-03-02 05:46:30] VERBOSE[3884] file.c: -- <SIP/2562-00000007> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[2013-03-02 05:46:32] VERBOSE[3884] file.c: -- <SIP/2562-00000007> Playing 'check-number-dial-again.gsm' (language 'en')
[2013-03-02 05:46:33] VERBOSE[3884] pbx.c: == Spawn extension (from-internal, 18506446200, 6) exited non-zero on 'SIP/2562-00000007'
[2013-03-02 05:46:33] VERBOSE[3884] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/2562-00000007", "") in new stack
[2013-03-02 05:46:33] VERBOSE[3884] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2562-00000007'
[Nexus 4 General] ttabbal:
What do your outbound routes look like? That's almost always the problem when you get that message.
[Nexus 4 General] easyxtarget:
Thank you, I didn't check the right boxes in the google voice module so outgoing wasn't configured at all.
[Nexus 4 General] Demonic240:
I may look at this at some point, though right now I'm enjoying the pbxes.org > css set up. (far simpler than all this it seems)
[Nexus 4 General] ttabbal:
With css, it keeps getting. ... stuck.. for lack of a better term. The sip stack process seems to stop responding occasionally and the only ways to get it running again that I have found are to reboot or use a root task killer to kill the sip stack process. Going into the app works, but attempts to connect, disconnect, make calls, all fail. Anyone else see that? If not I'll try reinstalling and configuring from scratch to make sure it's not bad data.
[Nexus 4 General] acegolfer:
I have experienced such zombie behavior in the past where a reboot was necessary. These days most of the issues get resolved when I force close the css from the app infor.

I admit that CSS is the weakest link in this setup but there are no better alternative sip clients that support the codecs I use.

I have been experimenting with iax client such as zoiper to overcome the sip issues. The problem, however, is lack of codec support.
[Nexus 4 General] swaggerwocky:
If anyone is having issues with missed calls using TCP 5060, it's a

conntrack issue according to the developer of CSipSimple

. motley was kind enough to disable the offending modules in his kernel, which can be found

here

.
[Nexus 4 General] Demonic240:
You should also be able to go to settings > disconnect and then relaunch css. I believe this method is better than fcing it.
[Nexus 4 General] ttabbal:
Doesn't work when in this mode for me. It doesn't happen often, but what I'm talking about is the app thinking the SIP stack is responding, but it's really not. Nothing happens to it. I can do disconnect, tap on the icon in the Accounts list, whatever, nothing changes. Killing it with a task killer is the only thing that works. And I have to kill the SIP stack process, not the main CSIPSimple one.
[Nexus 4 General] osi13:
How has SILK been working out for everyone? I'm thinking about installing PIAF on EC2 once I have a break from school coming up.
[Nexus 4 General] Demonic240:
Does PBXes.org support SILK?
[Nexus 4 General] acegolfer:
No.
[Nexus 4 General] acegolfer:
Latest module update has broken my setup. In half of my calls, there was no audio.

I decided to build the system from scratch, which took about 30 minutes.
[Nexus 4 General] Warbuff:
Im on purple. And last week when i initially set the server I was able to do both outgoing and incoming calls. I decided to start from scratch again on purple and now I am able to do outgoing calls over both TCP and UDP but I am unable to receive any calls. I added my google voice number to the inbound route under the DID field and selected the correct extension.

EDIT

Just did the same in Green and problem still occurs. I can dial out but I cannot receive any calls even with the proper inbound rules.

Note of Interest

I was attempting to set up Simonics but couldn't get it to work properly so I stopped using it. Do you guys think maybe since my account is still registered on his website that is why Im unable to receive any calls to my gv number. Im unable to delete the account because I used an application specific password which I stupidly did not write down.

Just tested on Groove ip. And I'm able to receive calls there.
[Nexus 4 General] totalnub911:
Have you been able to look at the asterisk console when this happens? When logged in, asterisk -rvvvvvvv

From there you should be able to see where things are going wrong.

Interestingly, I too seem to be suffering from inbound calls not going through. Everything looks good on the server side and the console shows "calling 701" but it never says "701 is ringing"

This may have to do with csipsimple and not asterisk, but it has been causing me trouble ever since I started. I've tried everything from piaf- green and purple on ec2 to incrediblepbx on a VM to incrediblepbx on my raspberry pi..... Nothing seems to consistently work and I'm getting a bit discouraged....

If anyone has seen this happen, could you please chime in?
[Nexus 4 General] acegolfer:
You can email Bill Simon (simonics creator) and he can manually delete your account even if you used app-specific password. But many people found simonics GVGW to be superior to Chan_motif. YMMV
[Nexus 4 General] acegolfer:
It has been noted several times in this thread that csipsimple-piaf connection is the weakest link.

I have experienced a similar behavior. In my case, when screen is off, CSS receives the call but it doesn't wake up the phone to ring. Once I unlock the screen, I noticed that there was a missed CSS call. OTOH, when the screen is on, CSS rings the phone normally.

To overcome this issue, I have GV forward calls to my Tmo number (in addition to gchat) so that it forces the phone to wake up. I just decline the Tmo phone and answer via CSS.
[Nexus 4 General] totalnub911:
Interesting workaround. Is this the only way? is it limited to CSS?
[Nexus 4 General] acegolfer:
1. I haven't tried other methods to fix. Somebody may chime in.

2. I only use CSS because it's the only free app with SILK codec.
[Nexus 4 General] swaggerwocky:
That's odd, I've never had any issue with CSS being unable to wake up my phone for an incoming call. For reference, I'm using CM10.1 linaro built by trailblazer + motley kernel b49 along with nightly CSS.
[Nexus 4 General] Warbuff:
Update on my situation. Emailed Simon and was able to remove my account from Simonics. I added the account back and this time was able to save everything successfully in piaf green. Everything working well here with silk12 codec. I'm able to receive calls with the screen off so far. Although this will require more testing.

I'm curious on your security setups. Did you guys set up the white list travelling man or some sort of other security? Btw thanks for your help acegolfer
[Nexus 4 General] totalnub911:
That's another thing I've run into, SILK doesn't seem to want to work. I've installed the codec on piaf-green and can verify that it is there by doing "core show codecs" but it doesn't show up in the web GUI as something I can enable and when I set css to use silk I get a rejected message when I try to place a call. Is there a step I'm missing?

I feel dumb having to ask all of these questions as I am very comfortable with Linux and know quite a bit...
[Nexus 4 General] acegolfer:
Go to Extensions in web GUI,

disallow=all

allow=silk12
[Nexus 4 General] totalnub911:
Figured this out right after I posted. Added silk12 to the allow line on my extension.

Still having trouble getting the phone to wake for calls... Doesn't even show call missed. I'm using simonics gw and have noticed that only one call at a time can go through.... Might have to go back to motif.

On another note, does anyone know how obitalk gets theirs to work even when you have gtalk signed into on multiple devices? Priority <1?
[Nexus 4 General] clintb:
That one was messing me up as well. Such a simple thing, but with so many options in the interface it can be a bit daunting.
[Nexus 4 General] Brozono:
No incoming calls

I have recently set up PIAF on EC2 and things were good up to testing incoming calls.

I can make calls from my phone to others and things work just fine. However when I test incoming calls nothing happens. Asterisk console doesn't show anything. No capture call coming in.

I originally had PBXES.org setup using callcentric forwarding for inbound. When I disabled callcentric I could see PBXES.org working. Then I though perhaps PBXES.org was answering the call for me. I deleted the routes and trunks from PBXES.org, but didn't see any improvement. Still the inbound call doesn't seem to hit PIAF.

I then tried Simonics, but failed to understand how to setup the trunk and routes. I have deleted this and tested again. Outbound is fine, but inbound still never gets to PIAF.

Can someone point me to some instructions on setting up Simonics in PIAF and if I should remove the Google Voice trunk (didn't see how to do that, I had tried to delete the routes and trunk, but then re-adding caused an error with number already used - I recreated the instance when this happened).

Can someone suggest some other debug I can try?

Thanks!
[Nexus 4 General] Warbuff:
http://nerdvittles.com/?p=832

These are the instructions I used to set up simonics. Before you start make sure that if your used the inbuilt google voice module in piaf to delete all the trunks it created. And then start the guide. After finishing the guide I had the same problem and want receiving any calls. What solved it for me was to log out of all google accounts on the computer.
[Nexus 4 General] totalnub911:
Make sure you sign out of chat on your phone. It will auto login unless you manually sign out.
[Nexus 4 General] pedxing:
acegolfer, I'm at my wits end. Have you encountered the problem on the N4 that causes alerts to not go through in a timely manner while on wifi when the device has been asleep for a while (10+ minutes)? I think this is the only thing killing me for this working reliably. What kernel/ROM combo are you using?

Here's my setup: I started with PIAF for Vmware (running asterisk 11.2.2). I've since upgraded to asterisk 11.3 rc 1 because it supposedly fixes the bug where the google trunk stops accepting calls after a random period of days. Because of this, my config is based on the config files only and I'm not using the PIAF web config tools. Other than that it's set up like your guide. I have g729 installed and I'm pleased with the quality. Incoming and outgoing calls are just fine with low latency on both wifi and 3g.

On the phone I have AOKP 4.2.2 build 5 and Franco.kernel r100. Running the latest dev build of csipsimple using tcp. Keep alives are set very low at 60 seconds. I also have the registration timeout set quite low at 300 seconds.

If the screen is on, any incoming call shows up and the ringtone goes off instantly. If the phone has been asleep for 5-10 minutes I get other results. Sometimes it answers but is slow (2nd or 3rd ring from the caller's side). Sometimes the phone wakes right when the calls is transferred to GV voicemail. Sometimes it does nothing and the call goes to voicemail. When this occurs, I can call a second time and it picks up immediately.

Franco has implemented the codeaurora patches and I've manually changed my wifi driver ini file to update the mbroadcast filter setting (to off) and the arp offload setting to 1 = on. Pings go through to the phone regardless of time asleep.

Any advice you can give me? What setup are you using on the phone side?
[Nexus 4 General] acegolfer:
I'm on stock rooted.

I posted my workaround for WiFi issue a few days ago.
[Nexus 4 General] pedxing:
Was that the one where you enabled it in GV to ring both gtalk and the nexus through the carrier at the same time? (I searched all of your posts to find it, I missed it somehow).

I've tried that in the past and it's pretty unusable. Even with the sleep issue the CSIP screen would pop up on top meaning if I took the time to flip to the regular dialer, decline the call and flip back to csip the call would be gone. If I didn't do this it would continue to ring the dialer and no audio would go through on CSIP (although I could hear the caller).

...so I just tried it again and it seems to be much better...maybe one of the many updates fixed that issue. I'll leave it like that for now and see how it goes. Although it's a bummer because I also was using that Tasker script to put the device into Airplane mode but with WIFI left on to save additional battery. But reliability trumps battery.

Speaking of battery...are you still using the setup here: http://xdaforums.com/showthread.php?p=38507470#post38507470

I'm still on port 5060 but have qualify off because I noticed the same thing you mentioned. I'll add compact headers today on the asterisk side...I have it on CSIP but it may not have done anything. Still learning asterisk.

Thanks for your help!
[Nexus 4 General] pedxing:
Played with is some more, and yeah, ringing both the sip and tmobile at the same time isn't working. If CSIP is in the foreground, I can accept the call and the tmobile call will terminiate automatically. However if the normal dialer is on top, it won't allow me to do anything while it rings...no access to the notification bar or the home button. I can only accept or reject. If I reject it, it terminates the CSIP call as well and goes to voicemail.

Do you handle it a different way?
[Nexus 4 General] acegolfer:
Strange. When both dialer and CSIP ring, I can reject the dialer and CSIP will continue to ring.

I don't think my settings are any special. I use ring group so that PIAF can ring 3 devices.
[Nexus 4 General] acegolfer:
I don't think you did anything wrong. I have seen this error in the past 1-2 times. Normally, this error goes away.
[Nexus 4 General] Brozono:
Still hasn't gone away, yet.... I'll try the Google Voice Trunk again.
[Nexus 4 General] pedxing:
I searched some more on this issue and found numerous issues made in the csip issues log for this... All with my same problem. No real resolution.

One work around I found is that the Google now ring is actually there in the dialer as well. At least in the AOKP sethyx dialer. By adding a ring target of "last app" it will flip you back to the csip answer screen. Oddly it must be last app and not the csip app itself. Otherwise you get the csip dialer activity.
[Nexus 4 General] Demonic240:
Not sure what I'm missing here.. I've added the Silk codecs and confirmed that they're all listed in "core show". I added the lines (from post #71) and did a graceful restart from the admin cli. Went to my extensions and allowed All codecs. Silk still does not show up under the SIP Settings (not sure if it's supposed to?) Currently have Silk8 set up as priority when connected to Wi-Fi, but it's still skipping it and going to g722.

Also having issue for not receiving calls (Still working on this, and I've already added the inbound route). I've tried signing out of google on my computer and phone, but I'm still not seeing it route to my server.
[Nexus 4 General] Brozono:
Update. After deleting the Simonic's account and signing out of Google+ and Talk on the phone, and disabling chat in Gmail I was finally able to get inbound calls to work. Thanks for all the help!

Now to get minimal echo and background noise for the otherside of the calls so my WAF goes up.

---------- Post added at 01:13 PM ---------- Previous post was at 01:11 PM ----------

I took me a combination of disabling Chat in Gmail in the browser, signing out of Google+ and Talk on the phone. I think the critical one ended up being Talk, but I haven't gone back and signed back into these to do an exhaustive test.

I am however able to have the Voice Chrome plugin running on the computer and receive inbound calls.
[Nexus 4 General] acegolfer:
1. I guess I didn't have to go through all that because I have a dedicated GV gmail account. It is very easy to move GV number from one account to another gmail account.

2. When I'm in a noisy place, I disable echo and lower the mic volume all the way down in csip. Just press the wrench button during a call. Or use an inline mic earphone.
[Nexus 4 General] Demonic240:
I have a secondary gmail account that I rarely use (my "professional" looking one). Could I just transfer my GV number from my regular account to it to avoid any headache?

Nice advice about the echo! Didn't know that would have an affect in noisy places.
[Nexus 4 General] Brozono:
Great point. I haven't thought about moving the GV number. And hadn't really looked into thinking it might be hard. So then do you use the Google Voice app on the phone though and if so don't log into that account from any of the other google services?

Thanks for the suggestion on CSIP configuration.
[Nexus 4 General] acegolfer:
Yes.
[Nexus 4 General] acegolfer:
I log in to my GV gmail account from 3 places.

1. Simonics for phone calls

2. Android GV app

3. Chrome browser GV extension

#2 and #3 are not phone services so they don't interfere with Simonics.
[Nexus 4 General] Demonic240:
So now I've got something weird going on.. I couldn't get inbound to work, so I create a new number with my second GMail account and connected it to my PIAF account. I removed my previous GMail account from PIAF and tested some calls. If I called my new number, it would work just fine and I'd receive an inbound call (horray!). So I then signed out of everything possible for my old number, disconnected my account from my phone and re-added it back to PIAF while removing the new number.

If I call my old number (which is associated with PIAF), I never receive a call. I checked and saw that PBXes.org was still receiving those so I deleted that account, but it didn't appear to help.

If I call my new number (which is no longer listed in PIAF), it still rings to me phone.... which is bloody odd. Should I have to restart PIAF for it to sync properly?

Migrated my number to my second account. Apparently google doesn't like making a call from one google number to another. Once I dialed from an actual phone I was able to receive calls.
[Nexus 4 General] Warbuff:
Had to shut off my server as I was seeing intrusion attempts from China. Anybody got any tips on banning an ip range for a whole country or using a white list.
[Nexus 4 General] bluespire:
The least you will be able to do is to whitelist every IP block that AT&T uses. You'll have to find out what that is. But the easiest way to start is to find out the IP address that AT&T is currently assigning to you, then plug that into domaintools.com. Domain Tools will give the CIDR block that that address belongs to. When you have that, go to your security group and for every port/port-range you have, add that CIDR address in the "Source". If you do this for everything, the only caveat will be that you will need to be tethered via your phone to use SSH (since it won't allow anything outside of that range. If you do this, it should be VERY secure. After that, you will have to find out if there are any other IP ranges that AT&T uses and start compiling them together.

---------- Post added at 10:15 AM ---------- Previous post was at 10:08 AM ----------

I found this, that might help.

http://www.howardforums.com/showthread.php/1572452-ATT-IP-address-blocks

AT&T is 32.0.0.0/8. You will get a NATed address, but Amazon will see the 32 address. Your only problem will be when you are on WiFi. Not sure how to fix that issue, since it changes so much more frequently and is more varied. I keep my phone off wifi the majority of the time, so it doesn't affect me.
[Nexus 4 General] Demonic240:
I keep seeing these unknown call log entries in my CDR report.. but I haven't made any call attempt today. Anyone have an idea?
[Nexus 4 General] acegolfer:
You will get a faster and better response from Piaf forum. It's very active.

Have you checked the log file?
[Nexus 4 General] Demonic240:
Are lots of people still using this? I haven't seen any comments in a while.
[Nexus 4 General] Warbuff:
I still use it
[Nexus 4 General] clintb:
I am. After some initial pains of configuration trying to sort out instructions between PIAF Green and Purple, the Amazon EC2 part, and moving my instance to N. California, I finally got it and it's just up and working smooth as silk.
[Nexus 4 General] bluespire:
In all honesty, that's how a great solution should be. ;)
[Nexus 4 General] acegolfer:
I have been using it for 2 months. Roughly once a week, I need to kill CSS and restart.