PBXes » Search » Search Results
Showing posts 1 to 20 of 39 results Pages (2): [1] 2 next »
Author Post
Thread: RE: Incoming call problems
cc

Replies: 6
Views: 14432

RE: Incoming call problems 29.01.2010 02:12 Forum: Bugs

Zitat:
Originally posted by telagente00
Is that why I see the two entries in call monitor, and my calls fail due to the incorrect one ?
If thats the case then how do I get rid of the false one.
This started around 8 hrs ago so its clearly not going to resolve itself quickly. I have tried deleting the trunks and recreating on PBXes to no avail.


Possibly related to what happened to me?
http://www1.pbxes.com/forum/thread.php?threadid=1259783717

I got zero support... but managed to get it working by changing the software release in personal data section.

good luck!

Thread: RE: www5 down again
cc

Replies: 10
Views: 19755

RE: www5 down again 27.01.2010 19:18 Forum: Bugs

all these problems are getting maddening! For once, www2 wasn't affected so I can breath a brief sigh of relief... but I'm sure issues are just around the corner :/

I really wish pbxes was more robust. I have a windows 95 box that has better uptime then pbxes smile (j/k...)

I'm considering rolling my own * box in a linode or something - but I'm too scared... haha...

Hopefully iptel eventually creates something that is truly fault tolerant.

Thread: RE: voip.ms: call forwarding?
cc

Replies: 7
Views: 31763

RE: voip.ms: call forwarding? 15.01.2010 08:06 Forum: Providers

ayking,

That worked great! How did you figure that out?

Anyway, thanks a lot! smile

-Rob

Thread: RE: voip.ms: call forwarding?
cc

Replies: 7
Views: 31763

RE: voip.ms: call forwarding? 12.01.2010 02:09 Forum: Providers

Hi ayking!

Are you saying you have two trunks setup for voip.ms? Thus you have two voip.ms subaccounts?

trunk1: voip.ms1
username: voip.ms username (subaccount 1)
password: voip.ms password
Register: yes

trunk2: voip.ms2
username:
password: password:username (subaccount 2)
Register: no

Is this right? Thanks a bunch for confirming that it can work! smile

Thread: RE: voip.ms: call forwarding?
cc

Replies: 7
Views: 31763

RE: voip.ms: call forwarding? 09.01.2010 08:45 Forum: Providers

Hi bobmats,

Sorry for taking so long to write back.

Like I said in my first post, I'm doing the password:username trick for voip.ms (using premium routes) and it doesn't work - but it does work with CWU. :/

The only way I can get the voip.ms trunk to work is when I fill in both the username and password field. Then, obviously, CID forwarding doesn't work - but at least the call completes.

Are there any other tricks for me to try? What would cause a ResetCDR from voip.ms? Any help would be greatly appreciated!

Thanks!

Thread: RE: Telephones Down!
cc

Replies: 31
Views: 60073

RE: Telephones Down! 05.01.2010 21:08 Forum: Bugs

i-p-tel,

Should we quit using www2? Every single week it's down for one reason or another.

Can you give us any metrics regarding uptime or number of outages for www2 vs www4?

Thanks!

Thread: RE: voip.ms: call forwarding?
cc

Replies: 7
Views: 31763

voip.ms: call forwarding? 04.01.2010 08:12 Forum: Providers

I'm currently using callwithus for call forwarding to my cell phone. I wanted to give voip.ms a try.

When I setup voip.ms like I do with CWU (ie: password:username in the password field, username field blank), it doesn't work - I get ResetCDR. I can't get it to work correctly.

Has anyone used voip.ms this way before?

Thanks!

Thread: RE: Telephones Down!
cc

Replies: 31
Views: 60073

RE: Telephones Down! 17.12.2009 20:18 Forum: Bugs

Zitat:
Originally posted by i-p-tel
We have verified you on "millc2".

To solve the ongoing problems on www2 we replace the server hardware and upgrade RAM very soon. This will result in another short offline period. If longer than 20 minutes accounts will be swapped to www4.


Is the server replacement happening today because www2 is down again. I really hope this solves the issues with www2 because it has been an awful experience to be on that server.

Thread: RE: Telephones Down!
cc

Replies: 31
Views: 60073

RE: Telephones Down! 15.12.2009 22:47 Forum: Bugs

Zitat:
Originally posted by i-p-tel
There was a short interruption on www2 at 3:30 am Pacific Time. After a grace period of 8 hours accounts were switched bach to it.

At the time you reported an error after 1:00 pm Pacific Time our monitoring system did not report any errors. This issue rather seems like a partial outage of Internet connectivity.


my account is still down. When I go to the status widget it says "Error loading configuration file variables.txt?aldope=33521".

All calls go to busy.

2009-12-15 13:45:07 "blah" <1858205xxxx> 120 alpha3.call­centric.com ext-local Busy 00:00:00

Also, I just wanted to point out that the initial failover to www4 worked - it was the switch back to www2 that has caused issues for me.

Thread: RE: Telephones Down!
cc

Replies: 31
Views: 60073

RE: Telephones Down! 15.12.2009 21:39 Forum: Bugs

Zitat:
Originally posted by millc
PBXes DOWN - Please Investigate


Same for me *sigh* So much for failover and redundancy.

Man, I'm using them for a 1 person shop and it's killing me with all their downtime. How do companies that rely on them for 10+ phones tolerate this amount of downtime. Anyone actually measure pbxes downtime for the past 3 months?

BTW, the status pages says everything is great.

Thread: every call routed to a PSTN results in 3 calls
cc

Replies: 0
Views: 6353

every call routed to a PSTN results in 3 calls 02.12.2009 20:55 Forum: Bugs

Final Update: I got it working finally. I had to go in to 'personal data' and change the software version to Newest, Saved, and then changed it back to Stable.

Thanks for all the help with this... customer service is #1 here at pbxes.com </sarcasm> unglücklich
========

pbxes: why is this being ignored? Ever since the outage 2 days ago, my account has been broken! Please please please pretty please look into this. What do I need to do? I wish I could understand why pbxes is so unreliable for me.

=======

My last post talked about ring groups - after more digging - I have discovered this has nothing to do with them - it's more general. When a call is routed to a PSTN extension, it results in 3 calls.

I'm using callwithus for outbound calls.

Here is pbxes call log:
2009-12-02 11:44:23 "Anonymous" <18584571111> 18582052222 /sip.call­withus.com CallWithUs from-internal-cont Dial 00:00:07
2009-12-02 11:44:23 "Anonymous" <18584571111> 18582052222 ­ CallWithUs from-internal-cont Dial 00:00:00
2009-12-02 11:44:23 "Anonymous" <18584571111> 18582052222 ­ CallWithUs from-internal-cont Dial 00:00:00
2009-12-02 11:44:22 "Anonymous" <18584571111> 101 pbxe­s.org ext-local Dial (00:00:07)

Here is callwithus call log:
1. 2009-12-02 11:44:31 18584571111 18582052222 usa proper 00:07 ANSWER STANDARD 0.0099 usd 0.0100 usd 60
2. 2009-12-02 11:44:31 18584571111 18582052222 usa proper 00:00 CANCEL STANDARD 0.0099 usd 0.0000 usd 60
3. 2009-12-02 11:44:31 18584571111 18582052222 usa proper 00:00 CANCEL STANDARD 0.0099 usd 0.0000 usd 60

Please check into what is causing this bug.

Update: It's still happening! Here is the detailed log from today - this is totally broken:
Dec 3 14:11:48 VERBOSE[10401] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Dec 3 14:11:48 VERBOSE[10401] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Dec 3 14:11:49 VERBOSE[22908] logger.c: -- Called 18582053333@from-internal/n
Dec 3 14:11:49 VERBOSE[22908] logger.c: -- Called 18582053333@from-internal/n
Dec 3 14:11:49 VERBOSE[22908] logger.c: -- Called 18582053333@from-internal/n
Dec 3 14:11:50 VERBOSE[22920] logger.c: We're at 216.75.41.112 port 46688
Dec 3 14:11:50 VERBOSE[22920] logger.c: Video is at 216.75.41.112 port 43358

Why are 3 calls being placed? Please give me a little support.

Thread: RE: www4 is down
cc

Replies: 5
Views: 14131

RE: www4 is down 22.11.2009 04:26 Forum: Bugs

i'm on www2 which gets redirected to www4 which apparently is also down - sweet! :/

Update: phone is now registered - most likely to www4.

How come we aren't getting email updates anymore when there is a service interruption?

Thread: RE: Pbx - locked, but now resolved and released
cc

Replies: 6
Views: 13098

RE: Pbx - locked, but now resolved and released 20.11.2009 08:04 Forum: Bugs

i-p-tel,

Is there a technical reason why things like this happen? How come pbxes gets confused and we (the users) have to come in and click the 'submit & start' button?

This hasn't actually happened to me (that I know of), but I'm just curious.

Thanks!

Thread: RE: ring group forwarding not working
cc

Replies: 7
Views: 21168

RE: ring group forwarding not working 17.11.2009 20:24 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

sorry, noob here - but maybe try turning on call waiting on all the extensions via pbxes? That way if one phone is busy it should continue to ring the others.

Thread: RE: Cancelation of account by mistake
cc

Replies: 9
Views: 19235

RE: Cancelation of account by mistake 27.10.2009 01:12 Forum: Miscellaneous

I'd have to agree - i-p-tel, your customer service is just plain awful regarding issues like this (although, it's pretty good regarding bugs/issues in the software)

I mean, you're making a customer very unhappy over a few euros - and I'm sure the other people reading this are thinking the same thing.

oh well, pbxes has no real competition - so we'll just grin and bear it... for now...

Thread: RE: pap2t - help - random hangups :(
cc

Replies: 9
Views: 32122

RE: pap2t - help - random hangups :( 20.10.2009 21:09 Forum: Terminal Equipment

Ok, on monday the problem presented itself again - loud dtmf done followed by one-way audio. Monday night I enabled the settings you recommended:
'FAX CED Detect Enable: No and FAX CNG Detect Enable: No'

And so far today (tuesday) has been good. I don't know why my wife is having such awful problems with this pap2t. I also changed all DTMF to inband to totally prevent the pap2t from processing anything - I hate this thing! We have a pretty stable DSL connection, so DTMF should be fine with inband.

I have placed an order for a A580IP phone for her and I will change the dtmf handling back to rfc2833 after I hook the phone up. Does 'auto' default to rfc2833?

Anyway, thanks for the help... I'll report back if the problem happens again with all the FAX detection settings disabled.

Thread: RE: pap2t - help - random hangups :(
cc

Replies: 9
Views: 32122

RE: pap2t - help - random hangups :( 17.10.2009 23:48 Forum: Terminal Equipment

Hi Diafora,

My understanding of 'talk off' is this:
When the pap2t is using an out of band DTMF like AVT, it has to analyze the voice stream and try to detect when someone presses a digit on the phone. If a persons voice matches the frequency of a DTMF tone, the pap2t wakes up and sends the tone across the internet causing a random beep on the other side of the conversation. There are lots of threads on dslreports about this and the pap2t in particular.

So, my question is: Of those changes I made, the switch from 'Auto' to 'rfc2833' probably wasn't the setting that fixed my dropped audio problem, right? I wish I knew what all those settings were for - such as 'FAX Process NSE'.

Thanks again.

Thread: RE: pap2t - help - random hangups :(
cc

Replies: 9
Views: 32122

RE: help - random hangups :( 17.10.2009 02:13 Forum: Terminal Equipment

Diafora,

You are truly amazing... I have not had a single dropped audio call in 2 days now!

The changes I made were:
* Set DTMF mode to rfc2833 on all trunks & extensions in pbxes
* DTMF Process INFO: No
* FAX Passthru Method: ReInvite (I think)
* FAX Process NSE: No
* All other suggestions I already had set

I would never have made those changes on the pap2t if you didn't suggest them. The only thing I didn't do was set g729 as the preferred codec.

Do you have any thoughts as to what change most likely was the 'fix'? The wife may be causing some talk-off on the pap2t - do you think it's safe to switch everything regarding DTMF to 'inband' - I have read that reduces talk-off - you agree?

Anyway, I just wanted to chime in and say thanks a million for those suggestions. I think I"m going to get an IP phone (A580ip maybe?) and kick the pap2t to the curb :/

Update: talkoff is pretty bad - especially when she says things like 'eye' or 'uh'. It appears to go in phases from a couple of times a minute to no problems...

Thread: RE: pap2t - help - random hangups :(
cc

Replies: 9
Views: 32122

help - random hangups :( 14.10.2009 23:04 Forum: Terminal Equipment

Hi,

I am attempting to track down what is happening with my calls.

My setup:
pap2t->router (tomato 1.25)->att dsl (bridge mode)->internet

At random times, from 30s to 30min to never, I get random OUTBOUND audio drops. The connection is still active, but outbound audio (from the pap2t to the PSTN) has stopped for some reason.

The caller ALWAYS complains about a loud DTMF/FAX tone when the audio drops. I can hear them, they can't hear me. This happens on both sent and received calls.

Do you have any idea what would cause a loud DTMF tone right before the outbound audio is dropped?

I currently have the pap2t in the DMZ so it's not port forwarding issue. I haven't tried STUN yet, but I read that that is generally unneeded.

i'm going to see if I can get the pap2t on a public IP address somehow to see if it's a router/firewall issue. I wish I had another IP phone to test with to rule out the pap2t.

Anyway, any insight would be much appreciated...

Thanks!

Update:
Looking through the logs I see this for the call that ended badly - sip.callwithus.com is ONLY used for outbound calling... the call in question was answered by the pap2t - so why is CWU hanging up?
code:

Oct 14 14:20:50 VERBOSE[6592] logger.c: -- SIP/cc-110-907e answered SIP/5068-081b57c8
Oct 14 14:20:50 VERBOSE[6592] logger.c: We're at 216.75.41.112 port 39122
Oct 14 14:20:50 VERBOSE[6592] logger.c: Video is at 216.75.41.112 port 44934 
Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding codec 0x4 (ulaw) to SDP
Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding codec 0x8 (alaw) to SDP
Oct 14 14:20:50 VERBOSE[6592] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Oct 14 14:20:50 VERBOSE[6599] chan_sip.c: Hangup call SIP/CallWithUs-fd3a, SIP callid [EMAIL]18c220f00cfa2c0e703f27d26d7906ee@sip.callwithus.com[/EMAIL]
Oct 14 14:24:00 VERBOSE[6592] chan_sip.c: Hangup call SIP/cc-110-907e, SIP callid 26d6cde41e76e83a4d1f83ba462a299d@216.75.41.112
Oct 14 14:24:00 VERBOSE[6592] chan_sip.c: Hangup call SIP/5068-081b57c8, SIP callid 22354b6b72aaaf95478806b30b19e6b1@204.11.192.27 

Thread: RE: Problem with incoming calls
cc

Replies: 7
Views: 27756

RE: Problem with incoming calls 05.10.2009 20:49 Forum: Providers

I am also having issues with incoming calls today - I just get dead air... I have temporarily forwarded the DID to my cell phone.

pbxes, please figure out what is going on - your reliability has not been very good lately.

I missed 11 calls, thankfully most were from the same person trying to get through. The problems started somewhere between 11:28am and 12:20pm PST.

update: seems to be working fine now...

Showing posts 1 to 20 of 39 results Pages (2): [1] 2 next »

Powered by Burning Board Lite 1.0.2 © 2001-2004 WoltLab GmbH
English Translation by Satelk