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Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br> As for the actual upgrade, is it possible to step through each

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

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Hello, I''m trying to test some cookie code with some integration tests, but I''m not sure if this is possible. Currently I''m creating a session (using open_session) running some code that sets my cookie and this works fine, the cookie is being set. Now cookies are supposed to exist across different sessions, so I try to create a new session (again with

From: Matthew Wilcox <mawilcox at microsoft.com> The sbitmap and the percpu_ida perform essentially the same task, allocating tags for commands. Since the sbitmap is more used than the percpu_ida, convert the percpu_ida users to the sbitmap API. Signed-off-by: Matthew Wilcox <mawilcox at microsoft.com> --- drivers/scsi/qla2xxx/qla_target.c | 16 ++++++-----

r> <h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010= ) </h2>With SIP 3.2.X firmware (available on the Polycom download site)=20 and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20 showing statuses of Ringing, Inuse and Online and one touch directed=20 call pickup. <br>On the asterisk side all that needs to be done is to add a hint

Greetings, I am working on getting mount.cifs version: 1.11-3.2.4 on debian to mount a share on a samba server Version 3.0.13-1.1-SUSE on SuSe. This was working on older debian systems, but upon upgrading some of the systems to Lenny I am now having trouble mounting shares. Again, this was working and I have smbfs installed on the systems (which is what I used before). The samba server is set

2010/2/17 Arnaud Quette > 2010/2/17 Arjen de Korte: > > Citeren Charles Lepple: > > > >> I wonder if there are any cross-compilation targets we could use to test > >> some of the word-size assumptions. Also, we could add in some static > >> analysis tools. > > FYI, I submitted NUT to the Coverity Scan program > (http://scan.coverity.com) last

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys

I am *still* unable to mount shares from a Ubuntu 10.04 server, using a Ubuntu 10.04 laptop. I totally re-formatted both my desktop and my laptop with Ubuntu 10.04 (so that they would be using the same version of Samba). I am using the exact same smb.conf for the 2 machines (less the share definitions, which exist only on the desktop, known as "workhorse"). wbinfo -u, wbinfo -g,

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : <Set> MD5Secret : <Not set>

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hello, I have an old table that handles sessions. the primary key is a field called session_id and is the actual session id like "8df838303ufdfu838" however when i do the following in the model: set_primary_key "session_id" and the following in the controller: @session_hash = { "session_id" => @session.session_id, "session_user_id" =>

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :