On Wed, 2009-11-18 at 10:32 -0600, Josh Patten wrote:
> It would appear I spoke too soon. Not 10 minutes after my last post it
> happened to one of my helpdesk people. Ranga, I submitted ticket
> http://track.sipfoundry.org/browse/XX-7059 with snapshots of my servers.
I just assigned it back to
It would appear I spoke too soon. Not 10 minutes after my last post it
happened to one of my helpdesk people. Ranga, I submitted ticket
http://track.sipfoundry.org/browse/XX-7059 with snapshots of my servers.
Josh Patten wrote:
> Wow, so the proverbial crap hit the fan yesterday and my Adtran TA
Wow, so the proverbial crap hit the fan yesterday and my Adtran TA908e
started doing crazy things, like dropping about 25% of the voice packets
(making faxing impossible and making voice calls very stuttery) and
randomly hanging calls up. A firmware update and a reboot didn't fix it,
so I imple
Yes, it's intermittent (the worst kind) Do you want me to create an
issue or send it directly to your email?
M. Ranganathan wrote:
> On Mon, Nov 16, 2009 at 3:21 PM, Josh Patten wrote:
>
>> OK, I had a user report their first dropped call of the day.
>> Details:
>> Caller 4328 dials into auto
On Mon, Nov 16, 2009 at 3:21 PM, Josh Patten wrote:
> OK, I had a user report their first dropped call of the day.
> Details:
> Caller 4328 dials into auto attendant 4310, then goes through the motions to
> get to the helpdesk. Call is directed-pickup'd by extension 4467 dialing
> *794694 (I chang
OK, I had a user report their first dropped call of the day.
Details:
Caller 4328 dials into auto attendant 4310, then goes through the
motions to get to the helpdesk. Call is directed-pickup'd by extension
4467 dialing *794694 (I changed it to *79 a while back for testing,
haven't changed it b
Correct.
The problem with Asterisk 1.6, FreeSWITCH, and YaTE is that they freak
out when dealing with REFER on an attended transfer (YaTE simply
doesn't support REFER). For the time being I was using Asterisk 1.6 to
handle REFER which was working in all cases except for when doing an
attended
But this is straight to your adtran and no freeswitch or asterisk media
gateway?
On Mon, Nov 16, 2009 at 11:09 AM, Josh Patten wrote:
> I am happy to report that with sipXbridge patch21 there have been no
> dropped calls or one way audio reported so far this morning (about 2
> hours worth of ca
I am happy to report that with sipXbridge patch21 there have been no
dropped calls or one way audio reported so far this morning (about 2
hours worth of calling).
I'll post and create a bug report if this changes.
M. Ranganathan wrote:
> On Sat, Nov 14, 2009 at 1:00 AM, Josh Patten wrote:
>
I will know monday morning if Patch21 will do the job. I just made a 1
hour an 10 minute test call from my google voice number with no voice
drop. We'll see how it goes
BTW I know EXACTLY what you mean about laborious and lengthy QA. I have
to do it all the time for new voice applications and pr
I have no doubt that it will work with freeswitch when running through
the patch. Same thing with YaTE.
I don't think my FreeSWITCH settings would be a good thing to post on
the wiki because I essentially turned off all SIP security settings in
an effort to get it working. YaTE, on the other ha
Thanks for the update Ranga.
Josh, if you do apply patch21.zip and have this working with freeswitch on
pfsense, please let the list know. I would suggest using a separate posting.
Perhaps a write-up for the wiki so others can follow as well if you have the
time and get that far, with an example c
On Sat, Nov 14, 2009 at 1:00 AM, Josh Patten wrote:
> It also appears that YaTE is the same way. that one was a little easier to
> set up, but it's the same old song and dance: REFER trips it up every time.
>
> I really wish sipXbridge was stable for me. Even with patch20 I drop to one
> way audio
tml
>
> Mike
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Friday, November 13, 2009 8:57 AM
> *To:* Picher, Michael
> *Cc:* Josh Patten; sipx-users@list.sipfoundry.org;
> gca...@franklinamerican.com
> *Subject:* Re: [sipx-users] Call Forwardi
ment.net]
> *Sent:* Friday, November 13, 2009 8:57 AM
> *To:* Picher, Michael
> *Cc:* Josh Patten; sipx-users@list.sipfoundry.org;
> gca...@franklinamerican.com
> *Subject:* Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk
> MediaGateway
>
> Good point. Never tried
-users@list.sipfoundry.org;
gca...@franklinamerican.com
Subject: Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk
MediaGateway
Good point. Never tried it, but once you get pfSense up and running (it
aint hard!), installing freeswitch is 2 clicks. I don;t know about
configuring it, but at
t; Sent: Thursday, November 12, 2009 1:27 PM
> To: sipx-users@list.sipfoundry.org; gca...@franklinamerican.com
> Subject: Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk
> MediaGateway
>
> Unfortunately there is no fix for this other than submitting a bug to
> Digium and ha
1:27 PM
To: sipx-users@list.sipfoundry.org; gca...@franklinamerican.com
Subject: Re: [sipx-users] Call Forwarding: Sipxecs with Asterisk
MediaGateway
Unfortunately there is no fix for this other than submitting a bug to
Digium and having it ignored. The SIP stack in asterisk is pretty
shoddy
18 matches
Mail list logo
