res_xmpp.c jabber socket read error Lancing Tennessee


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res_xmpp.c jabber socket read error Lancing, Tennessee

I'm using the Incredible PBX on ubuntu, and it has been a little flaky lately ( lots of options greyed out on the GUI, "sip show" type commands saying "what is The one trunk not connected says it's connected on the GV Motif trunk settings page, but the ringing does not actually ring thru to the pbx. I will switch back to 11. I did another reboot of Asterisk and now its all working again.

Has google given us the slip again? tm1000 (Andrew Nagy) 2015-07-18 19:17:26 UTC #11 dziny: they are not available as rpms. You were absolutely spot on.First time around the upgrade failed right in the middle of updating one of the components due to low memory. All rights reserved.

kill trees [HomeImprovement] by abrogard7282. It's not known whether people resolved the connection problem through various updates or it went away on its own (i.e. hmm may have just answered my question EDIT: Yes I was compiling without the right xmpp module. It happenswhen: - I answer a call on the Motif channel too fast.

Hearing a lot of OBI devices are impacted as well, but some are able to restore service after upgrading firmware. I did this on After doing so, I was able to connect without a problem, and made my first call from my laptop through to my cell phone! If it's true it just means updating to oauth2. The issue is the trunk though, not the OS.

A bad password doesn't suddenly happen, and why would you let Asterisk sit there for a long time without fixing the issue? i suspect this was the cause... SOLVED Google Voice XMPP Errors Discussion in 'Help' started by Robert.Thompson, Oct 18, 2014. I am getting something like this in the log when placing a call: {noformat} [Sep 18 08:05:54] VERBOSE[8542][C-00000000] netsock2.c: == Using SIP RTP CoSmark 5 [Sep 18 08:05:54] VERBOSE[9012][C-00000000] pbx.c: --

Newer Than: Search this thread only Search this forum only Display results as threads More... So in the begginning I discarded it as mundane tcp error. Don't think a2billing cares if it's 11.22 or 13.8.2 (though I could certainly be wrong about that!) #10 sirdotcom, May 7, 2016 Last edited: May 7, 2016 Will Longo Expand Check your credentials on that third account.Click to expand...

Now go to DIDLogic and and end this madness! Note, i just tested that config and got errors like you show, but also i got from google answer why it not working. I will try another version jump, to see what happens, but if anyone knows a fix for this I would love to know. theisgroup 2016-06-27 03:26:14 UTC #18 it looks like I only upgraded freepbx and my asterisk is way old.

Forum content is licensed under a Creative Commons Attribution-ShareAlike 4.0 International License. WB3FFV (Howard) 2014-11-12 07:21:55 UTC #5 As a follow up, I went into FPBX and disabled the GV/Motif module, and then made sure no trunks/routes were present that would use it. Go to admin > Asterisk Modules. Another Google life experience!

Sure is more reliable. It operates like it did from the beginning. isb (Isb) 2016-06-24 17:17:30 UTC #15 Thanks... i suspect this was the cause...

the solution for me was turn this on. #11 Will Longo, Jun 30, 2016 wardmundy likes this. I have tried enabling and disabling the XMPP module in FPBX, it didn't seem to change anything.. gmail asterisk share|improve this question asked Oct 3 '12 at 11:55 Bhavik Patel 354821 is @[email protected] correct? –arheops Oct 3 '12 at 15:03 ya it is correct.. Does WiFi traffic from one client to another travel via the access point?

erwabo 2016-06-15 16:22:08 UTC #6 Hmm, yeah im not 100% sure whats the issue. Mozilla - Distrusting New WoSign and StartCom Certificates [Security] by chachazz263. It seams very similar to what's being talked about.What actually fixed the issue? Any assistance would be greatly appreciated.

Additional information: I tried trunk, branches/11, but I can not confirm or deny if it is reproducible or not, because with these versions once call is established the voice is never Join them; it only takes a minute: Sign up how to configure gtalk and jabber in asterisk up vote 4 down vote favorite I set all configuration in asterisk and working It truly is a powerful command · actions · 2016-Jun-14 8:05 pm · RobThompsonCaution - Newbie AlertPremium Memberjoin:2012-02-14J8G 0C9

RobThompson to RonR Premium Member 2016-Jun-14 8:39 pm to RonR28 trunks!May I I mitigated this by issuing a Wait(2) before Answer() whenever i receive a google voice call Just happens randomly.

The errors I get have to do with OpenSSL. and if I understand at that point I can run asterisk 12, which should solve my issue asterisk 11.14 solved the google voice issue, freepbx distro Home Categories FAQ/Guidelines Terms The error message at face value seems to point to a TLS incompatibility, as though Google were enforcing a more strict TLS scheme and the client side has an older OpenSSL WB3FFV (Howard) 2014-11-12 07:12:03 UTC #4 Thanks Andrew, guess I can remove that one, as I think GV doesn't really even work with the PBX anymore, or I seem to recall

I have version 11.10.0 I'm running Incredible PIAF on CENTOS 6.4This is in my logs and repeats[2016-06-14 23:44:12] WARNING[1858] res_xmpp.c: Parsing failure: Hook returned an error.[2016-06-14 23:44:12] WARNING[1858] res_xmpp.c: JABBER: socket I do know its 11.4 jfinstrom (TheJames) 2016-06-15 16:34:08 UTC #7 oauth2 is not supported by asterisk upstream. I will switch back to 11. For what it is worth, I'll set up an ObiTalk account using the malfunctioning GV3 account just to see if that makes a difference - I'll post the outcome here after

However, outgoing calls (and perhaps incoming, but I am not sure) placed through Motif never get answered. If you can't provide the backtrace, then we'll close this for now as it's a situation no one should realistically run into. tm1000 (Andrew Nagy) 2014-11-12 07:08:32 UTC #3 This actually has nothing to do with the commercial FreePBX module called xmpp.