I have had bug issues with v3.0 ATA SIP code. The issues I've seen become
evident when using the backup SIP proxy AltGk. If the ATA is running off the
backup proxy because primary has failed, the ATA will never re-register with
the backup proxy if a call is in progress. It just keeps on trying to
register with the primary GkOrProxy. If the call is left up long enough, the
registration will expire from the backup proxy and then it won't be
registered anywhere. What's worse is that if you have session timer
configured, the call will drop once the registration ages out of the active
backup proxy.
Also seen issues when failed over to backup proxy where INVITEs are first
sent to the failed primary proxy and the not the backup proxy resulting in
an 8 - 10 second delay in INVITEs being sent to the active backup proxy.
Also seen gratuitous ARPs being sent from ATA every 5 seconds to itself.
All of these have been reported to Cisco TAC.
-----Original Message-----
From: Vitaly Nikolaev [mailto:vitaly@cifnet.com]
Sent: Wednesday, February 11, 2004 5:50 PM
To: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Cisco ATA adapter firmware upgrade
Hmm.. we found that 3.0 have best quality when u use g729. and it looks
fine and i did not noticed any bugs..
Can you please plese give me more info about those bugs ?
Thx
On Wed, 2004-02-11 at 09:41, Maxim Sobolev wrote:
> You need a Cisco TAC account in order to download fiwmware. As long as I
> know you can get one along with support contract from Cisco (either
> buindled with hardware, such as router or by bying separate one).
> Alternatively you can contact you hardware supplier - they should
> provide you with firmware.
>
> -Maxim
> P.S. I'd not recommend 3.0.0 - there are serious bugs.
> P.P.S. 1-year Cisco support contact for ATA186 costs only some $30, so
> that you can consider buying one. As long as I know this is cheapest
> legal way to get a TAC account, after all this account will be valid
> even after support contract has expired.
>
> Ashok Patel wrote:
>
> > Hi, all iptel.org users/members. I have question about
> > my cisco ATA 186 adapter. I want to upgrade my cisco
> > ata 186 adapter firmware from version 2.16.2 ata18x
> > (build 030829a) to version 3.0. atasip (build
> > 031210a). I am an end user and not reseller of this
> > adapter. I have no luck from cisco web site to
> > download the upgrade firmware. Does anybody know any
> > site (other than or besides cisco) from where I can
> > download the firmware upgrade image file or can anbody
> > send me the firmware upgrade image file (to my e-mail)
> > in order to upgrade my ATA adapter?
> >
> > -Thanks in advance,
> > Ashok Patel
> > ashokpatel(a)yahoo.com
> > ashokpatel(a)email.com
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Yahoo! Finance: Get your refund fast by filing online.
> > http://taxes.yahoo.com/filing.html
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
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He he.. Saw the NDA from sipquest.. Did'nt want to sign it myself..
- Atle
* Arnd Vehling <av(a)nethead.de> [040211 20:18]:
> Atle Samuelsen wrote:
> >Ok. Does the software from meranetworks work good?
>
> Hard to say :)
>
> One major problem is that one cant configure the sip listen
> port so were are running the mera linux binary of the SIP<>H323 gw
> on a bsd box. And this configuration does core-dump under certain
> circumstances. As you cant define the sip listen port for the proxy
> i cant really recommend meranetworks sip<>h323 proxy.
>
> Next sw we will test is the former siph323 open-source and now
> payware sip<>h323 proxy from sipquest. Ive the sipquest NDA
> allows me to post the testing results ill we post em here.
>
> Unluckily it seems like everyone in the SIP business wants a written NDA
> from you before they even tell you their fone number :)
>
> -- Arnd
>
>
Hey All,
I've been messing with this URI rewriting all day yesterday and today and I am still having some problems.
I am able to re-write the request URI okay, but I can't rewrite the To: and From: fields in the SIP packet. Is that possible, and if so, how do I do it?
I've tried using the subst function in the textops module. It doesn't seem to be re-writing all fields though. It still seems to only affect the request URI and not the To: or From: fields. Maybe I am doing something wrong?
I have attached my latest ser.cfg file for your viewing pleasure. :) I am trying to keep all requests that are between our local registered locations stripped to area code and phone num, and all requests that are to be relayed to the PSTN gateway need to have a +1 prefix added to them for e164 compatibility.
Any help would be VERY much appreciated.
Thanks,
Darren Nay - dnay(a)libertyisp.com
I want to rewrite URI in one of my modules. How can i do it....i know to do
it in ser.cfg file but not in my module.....
_________________________________________________________________
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I wnet ahead and added it and ran ldconfig:
[root@comm01 root]# ldconfig -v
/usr/local/lib:
libradiusclient.so.0 -> libradiusclient.so.0.0.1
I have the same error coming up - "libradiusclient.so.0 cannot be found"
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
-----Original Message-----
From: Edson Gellert Schubert [mailto:egschubert@ig.com.br]
Sent: Monday, February 09, 2004 6:42 PM
To: Morris, Scott; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Installing SER Radius problem
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Have You run "ldconfig" after installing the Radius Client?
Edson.
- ----- Original Message -----
From: "Morris, Scott" <MorrisS(a)orau.gov>
To: <serusers(a)lists.iptel.org>
Sent: Monday, February 09, 2004 9:04 PM
Subject: [Serusers] Installing SER Radius problem
> I am running Redhat ES 3. I have installed the SER 8.12 RPM. I am
> trying to install the ser-radius-0.8.12-0.i386.rpm, and am getting an
> error. It states that the libradiusclient.so.0 cannot be found. I
> installed the radius client, and I checked and libradiusclient.so.0
> exists in the /usr/local/lib directory.
>
> Any ideas?
>
> Scott Morris
> Enterprise Network Engineer
> DOE - ORAU / ORISE
> 865-576-4672
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
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=zgPI
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Ok. Does the software from meranetworks work good?
- Atle
* Arnd Vehling <av(a)nethead.de> [040211 19:11]:
> Hi,
>
> Vovidas siph323csgw _doesnt_ work as gateway between a SIP-Proxy and a
> H323 Gatekeeper. We are currenntly using a demo version of a russian
> (meranetworks) for terminating SIP into a H323 based system.
>
> We dont know about any Open-Source Solution for SIP<>H323 at the moment
> but we havent really tried asteriks yet. Several people said asterisk
> should be able to do it but no one could supply an example config
> yet.
>
> -- Arnd
>
Linux Redhat ES 3
SER ser-0.8.12-0.i386.rpm
SER RADIUS ser-radius-0.8.12-0.i386.rpm
RADIUS client http://www.mcs.de/~lf/radius
I installed per the "SIP Express Router RADIUS
HOWTO"
including running ldconfig -v
I am still getting the error on running the ser-radius-0.8.12-0.i386.rpm
Package Not Found
Unlocatable Package libradiusclient.so.0
Required by 'ser_radius','0.8.12', '0')
In my past testings, I have been able to get SER working, SER with MySQL
working, but not this implementation with RADIUS.
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
It all depends on the tarif of your provider - in my case, using
VoIPoGRPS is much more expensive than the GSM tarifs (if I call somebody
in Austria).
Klaus
Arnd Vehling wrote:
> Hello Klaus,
>
> thanks for sharing your experiences. I somehow expected to get
> miserable voice quality with GPRS and i know about the high latencies.
>
> But beeing able to use a GPRS IP Volume Tarif and using something
> like this http://www.linuxdevices.com/news/NS5033717348.html with
> a ported sip soft client is too tempting :)
>
> best regards,
>
> Arnd
>
>
Hi all,
I add user using serctl like:
#serctl add 123456 pwd123456 123456(a)iptel.org
But when I use check_from, it always return fail. And the from is like 123456 <sip:123456@iptel.org>.
I wonder what need to be add in datebase and how to add it.
Maybe I need add alias using "serctl alias add 123456 sip:123456@iptel.org" ?
Jimmy
