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Montano
12-23-2008, 08:19 PM
It's been a while since I was on a call for over 30 minutes, but today my line went to a fast busy @ the 30 minute mark. Was there a fix for this ? Something I need to change in my PAP2T ? I really dislike having to call Sirius Radio, mainly because it takes them forever to solve billing issues.

Montano
12-23-2008, 08:23 PM
Here's an old thread on the subject:

http://forums.voipo.com/showthread.php?t=659&highlight=minute

Another:

http://forums.voipo.com/showthread.php?t=690&highlight=minute

Montano
12-31-2008, 04:25 PM
I didn't realize there were so few BYOD users...... Hopefully someone will come up with a fix for the 30 minute call limit.

dswartz
12-31-2008, 05:03 PM
there is no fix, since this is per voipo policy (e.g. the customer HW has to support the requisite protocol, and the GS does, and your pap2t doesn't.) i'm not criticizing them, just the way it is. this is why i had to upgrade from asterisk 1.4 to 1.6 :)

quattrohead
01-01-2009, 09:21 AM
I spent over 2 hours with Dell's billing and customer service dept's with the 502 a few days back. Never dropped the call, quality great.
Could they fix the billing issue, no !^cking way.

dswartz
01-01-2009, 09:31 AM
I spent over 2 hours with Dell's billing and customer service dept's with the 502 a few days back. Never dropped the call, quality great.
Could they fix the billing issue, no !^cking way.

I'm confused! We already know the 502 works, no? Or are you hinting that he should ditch the pap2 for a 502?

Montano
01-01-2009, 10:35 AM
1/2 hour calls are not the norm for me, but it would be nice not to have to worry about the phone cutting off after 30 minutes. Is GS the only company that has the proper protocol ?

Xponder1
01-01-2009, 02:06 PM
I am using the device they shipped me that I received yesterday and its cutting calls off at exactly 29.5 minutes. I just sent a email to support. Odd because before I hooked up this device I was using X-Lite to make calls and was on the phone 90 minutes with no problem except a slight bit of noise on the line and the person on the other end if out of state saying they heard a echo of some kind (odd since I was using a headset).

dswartz
01-01-2009, 02:09 PM
I am using the device they shipped me that I received yesterday and its cutting calls off at exactly 29.5 minutes. I just sent a email to support. Odd because before I hooked up this device I was using X-Lite to make calls and was on the phone 90 minutes with no problem except a slight bit of noise on the line and the person on the other end if out of state saying they heard a echo of some kind (odd since I was using a headset).

Hmmm, open a ticket I guess. Sounds like it isn't enabling rfc4028 :( Why the softphone works is another question...

Xponder1
01-01-2009, 02:43 PM
Hmmm, open a ticket I guess. Sounds like it isn't enabling rfc4028 :( Why the softphone works is another question...

Well the softphone did work and it was not just once. I checked my call log and there are at least 5 calls over 30 minutes from the softphone. I have opened a ticket.

Something else weird I noticed last night. The mac address printed on this adapter is off by exactly one digit. The last digit was off by one number.

dswartz
01-01-2009, 02:46 PM
Well the softphone did work and it was not just once. I checked my call log and there are at least 5 calls over 30 minutes from the softphone. I have opened a ticket.

Something else weird I noticed last night. The mac address printed on this adapter is off by exactly one digit. The last digit was off by one number.

Well, half the mystery is solved; from what I can find online, x-lite implements the session timer feature :)

Xponder1
01-01-2009, 02:58 PM
Well, half the mystery is solved; from what I can find online, x-lite implements the session timer feature :)

Dunno but my wife will never agree to keep this if it cuts her calls off at 29.5 minutes. She can talk forever....

voxabox
01-01-2009, 03:01 PM
I am using the device they shipped me that I received yesterday and its cutting calls off at exactly 29.5 minutes. I just sent a email to support. Odd because before I hooked up this device I was using X-Lite to make calls and was on the phone 90 minutes with no problem except a slight bit of noise on the line and the person on the other end if out of state saying they heard a echo of some kind (odd since I was using a headset).
with VOIPo supplied ATA (HT-502/HT-286), you should not experience the 30 min. cut off

I see some possibilities:


your ATA is improperly provisioned
your ATA is not been automatically provisioned by VOIPo

Xponder1
01-01-2009, 03:13 PM
with VOIPo supplied ATA (HT-502/HT-286), you should not experience the 30 min. cut off

I see some possibilities:


your ATA is improperly provisioned
your ATA is not been automatically provisioned by VOIPo


I think #1 is the answer. Perhaps because I was using the softphone before I got the ATA.

dswartz
01-01-2009, 03:15 PM
Dunno but my wife will never agree to keep this if it cuts her calls off at 29.5 minutes. She can talk forever....

Try this: "Honey, if you're going to talk for more than 1/2 hour, you need to use the softphone! (LOL)

burris
01-01-2009, 03:19 PM
These are the settings on my 502 relating to call timing. Don't know if this is relevant. All of mine are set to no..

Caller Request Timer: No Yes (Request for timer when making outbound calls)
Callee Request Timer: No Yes (When caller supports timer but did not request one)
Force Timer: No Yes (Use timer even when remote party does not support)

voxabox
01-01-2009, 09:21 PM
These are the settings on my 502 relating to call timing. Don't know if this is relevant. All of mine are set to no..

Caller Request Timer: No Yes (Request for timer when making outbound calls)
Callee Request Timer: No Yes (When caller supports timer but did not request one)
Force Timer: No Yes (Use timer even when remote party does not support)
AFIAK, these settings should do the trick (for VOIPo openser configuration)

Xponder1
01-01-2009, 10:44 PM
AFIAK, these settings should do the trick (for VOIPo openser configuration)

I can login to this things web control panel and see the basic settings but any of the other tabs it tells me I have the wrong password. How do I check those settings?

burris
01-02-2009, 04:37 AM
I can login to this things web control panel and see the basic settings but any of the other tabs it tells me I have the wrong password. How do I check those settings?

Call or put in a ticket to have support check it for you.

Xponder1
01-02-2009, 10:55 AM
I already have a ticket submitted. I received this update today.

We'll have an engineer see why you are experiencing this problem, and get back to you once we have a resolution. In the meantime, let us know of any updates, if any, on your end.

Montano
01-12-2009, 05:10 PM
I'm very happy to report that I made a 45 minute call today !! No more 30 minute cutoff :)

Is Voipo a great company or what !?!?

voxabox
01-12-2009, 05:18 PM
Good news, but
Need to further elaborate on equipment, call, server, etc

Xponder1
01-12-2009, 05:19 PM
I'm very happy to report that I made a 45 minute call today !! No more 30 minute cutoff :)

Is Voipo a great company or what !?!?

Yes they are. I think the service is great.
I can not wait to see what happens next. I have had the service less than a month and despite a few minor problems I am really happy with the product. I can not wait to see what happens next. Even in my short time here I have seen things continue to change for the better.

Thank you Tim and all the staff for your hard work and dedication.

Montano
01-12-2009, 05:24 PM
Good news, but
Need to further elaborate on equipment, call, server, etc

PAP2T, sip.voipwelcome, local call.

zcnkac
01-12-2009, 05:50 PM
Session timers are enabled only on incoming calls. Please test incoming calls to be sure of the cutoff.

I use Asterisk 1.4, and I manually backported session timer support from the 1.6 branch (digium bug id 10665). If people are interested, I can provide the compiled chan_sip.so and also the merged source file. Overwriting chan_sip.so in /usr/lib/asterisk/modules/ is enough (a restart of asterisk or the system is not required).

Montano
01-12-2009, 05:58 PM
I must disagree. All the calls I got cut off on, were initiated by me.

zcnkac
01-12-2009, 06:12 PM
Outgoing calls cutting off might have been because of something else. I have since switched to central01. Before that, outgoing calls would drop sometimes because asterisk could not "re-authenticate". It might also be because I was using codeblue in fromdomain.

I can definitely say that RFC 4028 session-timers are being enabled only for incoming calls.

dswartz
01-12-2009, 06:13 PM
Outgoing calls cutting off might have been because of something else. I have since switched to central01. Before that, outgoing calls would drop sometimes because asterisk could not "re-authenticate". It might also be because I was using codeblue in fromdomain.

I can definitely say that RFC 4028 session-timers are being enabled only for incoming calls.

Hmmm, that's weird. I could swear I had a number of calls dropped at the 30 minute mark when I was calling other people (that was on asterisk 1.4 before I moved to 1.6).

voxabox
01-12-2009, 07:19 PM
PAP2T, sip.voipwelcome, local call.

Local call does not count. This is voip...
Is it sip to sip or sip to pots?

voxabox
01-12-2009, 07:27 PM
Guys, read the rfc again
Sst is to prevent zombies

zcnkac
01-12-2009, 08:01 PM
Here are my asterisk logs for an outgoing and incoming call. I am using central01 on asterisk 1.4.21.2.

[root@pbx /etc/asterisk]# chans
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
192.168.x.x 1001 31c81cc17b9 00102/00000 0x4 (ulaw) No Tx: ACK
67.228.251.106 408XXXXXXX 84017861_11 00101/25111 0x4 (ulaw) No Rx: ACK


[root@pbx /etc/asterisk]# asterisk -rx sip show channel 84017861_11

* SIP Call
Curr. trans. direction: Incoming
Owner channel ID: SIP/408XXXXXXX-xxx
Joint Codec Capability: 4
Format: 0x4 (ulaw)
MaxCallBR: 384 kbps
Caller-ID: 408XXXXXXX
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
DTMF Mode: rfc2833
SIP Options: timer
Session-Timer: Active
S-Timer Interval: 1800
S-Timer Refresher: uac
S-Timer Expirys: 0
S-Timer Sched Id: 363759
S-Timer Peer Sts: Active
S-Timer Cached Min-SE: 90
S-Timer Cached SE: 1800
S-Timer Cached Ref: uac
S-Timer Cached Mode: Accept

[root@pbx /etc/asterisk]# asterisk -rx "sip show channels"
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
67.228.251.106 1408XXXX 5e5ba777527 00103/00000 0x4 (ulaw) No Tx: ACK
192.168.x.x 1001 NDI5N2I0OTE 00101/00002 0x4 (ulaw) No Rx: ACK


[root@pbx /etc/asterisk]# asterisk -rx "sip show channel 5e5ba777527"

* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 5e5ba777527e8db43f7aa42836e8bc48@central01.voipwel come.com
Owner channel ID: SIP/voipo-out-xxx
Joint Codec Capability: 4
Format: 0x4 (ulaw)
SIP User agent: FreeSWITCH-mod_sofia/1.0.trunk-11100
Username: 1408XXXXXXX
Peername: voipo-out
Original uri: sip:mod_sofia@174.132.131.133:5060
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive

Montano
01-13-2009, 08:11 AM
Local call does not count. This is voip...
Is it sip to sip or sip to pots?

Do we really know for sure anymore ? People calling me assume it's a POTS number.......

I did not dial a sip address, so do we assume it went to POTS ?

VOIPoTim
01-13-2009, 10:00 AM
No one is still seeing the problem and is just trying to better understand the technicalities of it, right?

zcnkac
01-13-2009, 03:33 PM
Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.

Better yet, session timers can now be enabled both on outgoing and incoming. It's optional, so adapters like PAP that don't support timers won't break but for others like grandstream/asterisk 1.6, it can be used. Sweet ! It's possible timers on outgoing calls is a new thing on central01. For asterisk users who want to try this out, add a session-timers=originate to your voipo peer settings.

dswartz
01-13-2009, 03:41 PM
Yeah, I had to tweak asterisk 1.6 that way when I first converted, to avoid having my calls chopped off :)

scott2020
01-13-2009, 04:39 PM
Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.



Very interesting. Is this just a fluke or did Tim or someone from VOIPo come out and say they definitely have taken the timers out of the mix? In the past I would get hit or miss 30 minute cutoffs with my Asterisk box. Now I am tempted to go back.

But, my GS 502 is working so well I don't really want to mess with it!

voxabox
01-28-2009, 02:41 PM
BYOD'ers,
I'm happy to report that that outbound calls (to plain POT numbers) have been verified working for 30+ minute durations - thanks to the wife;)
ATA: SPA-1001, 3.1.19(SE), behind router with STUN, NAT keep-alive, ports: 5062,16384-16482 forwarded

02/13/09, inbound calls have been working for more than 30+ mins.
Last time I checked to call log, my wife answered "her call" and talked to her friend for 80+ mins

Cheers,
-v

scott2020
01-28-2009, 04:33 PM
Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.


I'd still like the official word from VOIPo on whether or not the session timers are in fact gone, or is it a coincidence that some BYOD devices are working past 30 minutes. When I was running Asterisk 1.4 I would sometimes have calls go longer than 30 minutes and sometimes not.

Scott

bubbanc
02-11-2009, 09:11 AM
Did anyone get an official response on this?

scott2020
02-11-2009, 10:48 AM
Not that I have heard, but with the Linksys PAP2's they suffered from the same session timer problem. In order to get those to work, from what I understand, VOIPo had to turn off the session timers. That is how I understand it but I don't know for sure and didn't get an official answer yet.
Scott

bubbanc
02-11-2009, 10:49 AM
Since Voipo is now shipping PAP2T's because of shortage of GS devices, you would think they would have disabled the timers.

Xponder1
02-11-2009, 10:58 AM
Tim said it was taken care of. Lets just take his word for it. ;)

Smiles
02-11-2009, 11:31 PM
I received one of the recently shipped PAP2Ts. I have received incoming calls longer than 30 minutes.

scott2020
02-12-2009, 08:44 AM
Tim said it was taken care of. Lets just take his word for it. ;)

No problem, I must have missed where he said specifically the 30 minute session timers were removed. When I was using Asterisk sometimes I would get cut off at 30 and sometimes not, so I was wondering if it was more coincidence or if the timers were indeed taken off.

scott2020
02-15-2009, 11:48 AM
I moved everything back over to my Asterisk box, since people have had good luck with it. I am running Asterisk 1.4, and I was talking with my brother this morning. At exactly the 30 minute mark, my call got cut off. I got a fast busy on my end. So I don't know what people are doing to get around it or whatever, but I got cut off at exactly 30 minutes. This makes me think the session timers are on somewhere and that is why I wanted an official word whether they were enabled, disabled for some, or disabled across the board. My other trunks always work over 30 minutes so I'm not sure if it is on my end or not.

Edit: I called him back, and talked to him for 35 minutes without getting cut off. So I have no idea what is going on.

Scott

dswartz
02-15-2009, 12:21 PM
Was the cutoff call initiated by you or him? I remember speculation that which end originated it mattered as far as the timers being on or not. I got irritated enough several months ago to roll out asterisk 1.6.0 and it's been stable for me.

scott2020
02-15-2009, 12:32 PM
I called out to him both times, which makes it even more frustrating. I am on the phone with my dad, who called me, at the 32 minute mark and no cutoff so far. Seems very random.

dswartz
02-15-2009, 01:06 PM
Sorry for your pain :( All I can suggest is switching to 1.6 :(

zcnkac
02-15-2009, 06:05 PM
I patched 1.4.21.2's chan_sip with session timers. I have been using it for the last few weeks without problems. I can post the modified chan_sip.c and chan_sip.so ...

scott2020
02-15-2009, 06:15 PM
I don't get it. The timers are either on or off for everyone I would think. Sometimes I get cut at 30 minutes and sometimes not. I am lost.

dswartz
02-15-2009, 07:24 PM
Well, there may be a bug where it is not always getting set. On 1.4 I had the same experience as you (sometimes yes and sometimes no.) With 1.6 I never do.

scott2020
02-16-2009, 09:51 AM
That is an option I suppose. I can't believe no one else has this problem, especially since the talk was the 30 minute timers were taken away by VOIPo. There must me someone else here running Asterisk 1.4. Maybe they just don't talk as long as I do!

fisamo
02-16-2009, 04:17 PM
I run Asterisk 1.4, but calls on the line are generally short. For now, it's mainly my 'hobby' line, because I got in on the Beta and have good pricing on the line. :) However, as my girls get older, the line will definitely start getting heavy use. :eek:

In the meantime, I'll have to think of a good test number to call that would keep the call open without me having to do anything. :) Any ideas?

Edit: (It may be obvious, but I don't want to burn cell minutes or tie up my regular house line, and I'd like the call to be off-voipo network...)

bubbanc
02-17-2009, 11:03 AM
Call Dell tech support, if they come on quickly tell them you have to find your service tag number and to give you a minute or perhaps an IRS hot-line.

scott2020
02-17-2009, 04:11 PM
I know some other VOIP providers that would keep me on hold until I was old and grey, but I won't mention them here! ;) Dell seemed to be good for that when I called them too.

Anyway, my wife talked to her mom for 55 minutes without getting cut off, so it is very random. Perhaps it is something from an upstream provider or something? Depending on the call route, maybe one of VOIPo's providers does something with the session timers.

Somewhere in this forum I think I remember a patch someone created, a backport from Asterisk 1.6 to 1.4 to support the session timers. I'll dig for that.

zcnkac
02-17-2009, 07:58 PM
I have attached a zip file that contains the modified chan_sip.c and the compiled chan_sip.so, which supports session timers. These are for asterisk version 1.4.21.2.

!!! Backup the files /usr/lib/asterisk/modules/chan_sip.so and /usr/src/asterisk/channels/chan_sip.c !!!

Since there is a 97 KB limitation on attachments I have split chan_sip.c into two volumes. To compile it, overwrite /usr/src/asterisk/channels/chan_sip.c with the one from the archive and do a 'make all' in the /usr/src/asterisk directory. This will create a chan_sip.so in the /usr/src/asterisk/channels directory. Next do an 'amportal stop'. Overwrite /usr/lib/asterisk/modules/chan_sip.so with /usr/src/asterisk/channels/chan_sip.so. Do a 'amportal start' ...

For configuring this, disable timers globally by adding the line
session-timers=refuse
to /etc/asterisk/sip_general_custom.conf. Enable timers for voipo, by adding
session-timers=accept (or session-timers=originate) and session-refresher=uac
to PEER Details of the voipo trunk in FreePBX.

I have been using this for the last few weeks without any issue. YMMV. Use at your own risk.



Somewhere in this forum I think I remember a patch someone created, a backport from Asterisk 1.6 to 1.4 to support the session timers. I'll dig for that.

scott2020
02-17-2009, 08:25 PM
Great, thanks! I'll have to give that a try.

PS

Seems like 97k is pretty small for the attachment limit.