Re: [Asterisk-Users] cisco phones problems

[Greg Oliver]

use the qualify= syntax in your sip.conf and make sure it exceeds the
latency between the phones and asterisk server in ms.

-Greg

On Wed, 2005-09-28 at 16:17 -0700, Edwin Lam wrote:
> hi folks.
> 
> we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and
> we start having problems of dropping calls (actually the calls wasn't dropped
> it just the sound was muted for about 5-10 seconds, but most users will think
> the call dropped and hangup/redial). i've check the console output.
> there was a lot of messages like the following:
> 
> Sep 28 15:00:49 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is 
> now TOO LAGGED!
> Sep 28 15:00:59 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is 
> now REACHABLE!
> Sep 28 15:01:08 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3201' is 
> now TOO LAGGED!
> Sep 28 15:01:18 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is 
> now REACHABLE!
> Sep 28 15:04:01 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is 
> now TOO LAGGED!
> Sep 28 15:04:11 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is 
> now REACHABLE!
> Sep 28 15:05:22 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '3201' 
> is 
> now UNREACHABLE!
> Sep 28 15:05:32 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is 
> now REACHABLE!
> Sep 28 15:06:23 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '4881' 
> is 
> now UNREACHABLE!
> Sep 28 15:06:33 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '4881' is 
> now REACHABLE!
> 
> we're running Asterisk 1.0.9 on Debian Sarge w/ custom kernel 2.6.12, strange
> thing is the system works fine with 2 Cisco phones & 8 Grandstream phone
> before, until i replaced the Grandstreams with Ciscos. the following is
> typical setting in sip.conf:
> 
> [1234]
> context=default
> type=friend
> host=dynamic
> username=1234
> secret=test123
> mailbox=1234
> callerid="John Smith" <1234>
> qualify=yes
> dtmfmode=rfc2833
> 
> any thoughts why this is happening?
> 
> 

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Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice

[Greg Oliver]

IMHO - you should not use price and quality in the same sentence for BV.

On Mon, 2005-09-26 at 15:20 -0400, Jason Schafer wrote:
> I'm relatively new to the whole VOIP game, here's what I want to do.  I 
> am using VOIPJet for all of the outbound calls on our AAH box.  I have 
> one landline that I would like to busy forward to an inbound VOIP 
> number.  Broadvoice was recommended to me for price and quality.
> 
> Can anyone make a suggestion for a good VOIP Provider for my inbound 
> requirement?  The bulk of my inbound calls will come in on the land 
> line, but I would also like the leverage the group/conference feature in 
> AAH (8+ext) and an inbound SIP seems to be a good answer for having a 
> couple of different people call in at once (three people call the SIP 
> number).
> 
> Jason
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Re: [Asterisk-Users] Asterisk to CCM

[Greg Oliver]

Are you using CCM to operate your gateway with MGCP?  If so, I had to
change the default timers under CCM advanced setup for "Media exchange
timers" or the call was timing out at 4 seconds.  If the setup was
complete prior, it worked fine, but after 4 seconds q.931 from CCM would
tear down the call..

On Mon, 2005-09-26 at 14:14 -0300, Arnaldo M. Pereira wrote:
> Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk%
> 20Cisco%20CallManager%20Integration ?
> 
> I've followed these steps and I can make calls from a CCM client to
> Asterisk, but the end point at the Asterisk side can't hear any audio.
> 
> On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote:
> > I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 
> > and vice versa. I have a SIP trunk setup in CCM and I also have an entry in 
> > my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM 
> > I keep getting:
> > 
> >  SIP/10.0.0.1-9c18 is circuit-busy
> >   == Everyone is busy/congested at this time (1:0/1/0)
> > -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 
> > 10.0.0.1
> > 
> > I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. 
> > I can send calls to it and they complete, but when I point the route 
> > pattern to Asterisk it fails immediatly. Any suggestions?
> > 
> > Thanks,
> > Brian
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Re: [Asterisk-Users] Cisco 7206 and Sample configs (Newbie)

[Greg Oliver]

Using 'dial-peer voice  voip' and 'dial-peer voice  pots' is
what you are looking for on the 7206..

Use sip, and just use dial(SIP/[EMAIL PROTECTED]) to make calls
go out..

search on cisco for dial-peer syntax and requirements - plenty of
examples out there.

-Greg

On Mon, 2005-08-08 at 12:48 -0400, Ronnie Tartar wrote:
> Not really sure, I'd like to set it up as SIP trunks, we currently have call 
> manager on some other servers but may replace (too expensive).   I'm just 
> kind of learning some of these VOIP terms, but I'm thinking we could 
> probably do this.
> 
> Also, they are T-1's coming into the 7206's, not T3's.  I would like to 
> leave the call manager out of it alltogether.
> 
> 
> 
> - Original Message - 
> From: "Lull, Rick" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Sent: Monday, August 08, 2005 10:08
> Subject: RE: [Asterisk-Users] Cisco 7206 and Sample configs (Newbie)
> 
> 
> >
> > Are you going to run Call Manager Express on that router or are you going 
> > to
> > use it in a different manner?
> >
> > I've had so-so luck making CME talk to * and back for all functions, but 
> > it
> > can work.
> >
> > Rick
> >
> > -Original Message-
> > From: Ronnie Tartar [mailto:[EMAIL PROTECTED]
> > Sent: Saturday, August 06, 2005 5:13 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Cisco 7206 and Sample configs (Newbie)
> >
> > Newbie to Asterisk
> >
> > I've been looking around for a little while, can't seem to find some 
> > sample
> > configs for using a Cisco 7206 as a gateway.  The below link is an initial
> > plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/
> > IPCC / IVR setup.  We currently have all of the hardware below.  Just take 
> > a
> > peak and see if there is anything that is off base.  I don't know If I 
> > will
> > be able to use the vg248 but would like to.  We have a call center that is
> > currently dormant but are considering opening it back up (approx 200 
> > seats).
> > Also looking for ways to cluster or make it highly available.  I looked
> > around, not a whole lot of info on this.
> >
> > Thanks in advance.
> >
> >
> >
> > http://www.designatedsystems.com/dynlink.jsp?link_id=63
> >
> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> > 
> > 
> >
> > The information in this communication is intended to be confidential to 
> > the Individual(s) and/or Entity to whom it is addressed.
> > It may contain information of a Privileged and/or Confidential nature, 
> > which is subject to Federal and/or State privacy regulations.
> > In the event that you are not the intended recipient or the agent of the 
> > intended recipient, do not copy or use the information
> > contained within this communication, or allow it to be read, copied or 
> > utilized in any manner, by any other person(s).  Should
> > this communication be received in error, please notify the sender 
> > immediately either by response e-mail or by phone,
> > and permanently delete the original e-mail, attachment(s), and any copies.
> >
> > ___
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> > 
> 
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Re: [Asterisk-Users] Failover question

[Greg Oliver]

> A hot thing in databases right now are clusters.
> Has anyone setup a linux cluster and installed asterisk on it?

I assume you mean replication - and yes, we use that environment here
and it works well in most circumstances with HA running the failover..

We do not do much call processing, but use Asterisk mainly for our 3rd
party software to do applications that are not possible with
CallManager..

Replicated databases and HA failover is OK for us there because we do
not have any phones registered (they are always registered to CCM).

-Greg

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Re: [Asterisk-Users] Cisco Voip Question

[Greg Oliver]

Router#conf term
Router(config)#voice class codec 99
Router(config-class)#codec preference 1 g711ulaw
Router(config-class)#codec preference 2 g729br8
Router(config-class)#codec preference 3 g729r8
Router(config-class)#end 

Router(config)#dial-peer voice 2000 voip
Router(config-dial-peer)#voice-class codec 99
Router(config-dial-peer)#^Z

-Greg


On Thu, 2005-06-30 at 10:22 -0400, Brian C. Fertig wrote:
> Does anyone in here know how to setup auto negotiation between g729 and
> g711ulaw on 
> a cisco 5400?   I would imagine it would be the same on a 3660.  
> 
> The problem I am having is natively the call is setup for g729 however
> when the call is transferred 
> to voicemail it uses ULAW so when the cisco tries to connect to the
> voice mail I get a SIP error
> that the codec couldn't be negotiated.  I need a little in sight on how
> to setup the dial peer or 
> something in the global config for the router.
> 
> 
> TIA
> 
> ..o---o.
> Brian Fertig
> NOC/Network Engineer
> Planet Telecom, Inc.
> Tampa, FL Office
> 
> 
> 
> 
> 
> This email was scanned by:  Mcafee GroupShield
>  CONFIDENTIAL DISCLAMER 
> All information provided in this email is considered confidential
> and proprietary of Planet Telecom, Inc. and Telecenter Inc.
> Use of this information by anyone other than the recipient or 
> sender will be considered in breach of agreement.
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RE: [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth?

[Greg Oliver]

You may also want to do some packet captures when you experience the
problem for both the Linksys and the Vonage ATA to see what they do
differently..

-Greg


On Wed, 2005-06-29 at 17:59 +0200, Marcel van Kaam, Fonetica wrote:
> I have my systems running on ulaw, alaw or GSM. No other codecs. Myself I
> even prefer the ulaw because of the quality.
> 
> I will look tomorrow a little bit further in the Linksys as I have 2 of them
> here to test and so far I am very happy with them. 
> I will play a bit around with the settings and let you know tomorrow or I
> founded some things to improve. 
> 
> Marcel 
>  
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding
> Sent: woensdag 29 juni 2005 16:45
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings
> forperformanceandlowbandwidth?
> 
> I have indeed already done so - I use G729 quite a bit since I travel alot 
> in adverse conditions.  Interesting thing is, I can get less choppy audio 
> sometimes from my Vonage device using (what I suspect to be) Ulaw, while 
> either ulaw or G729 will still give choppy response at that moment from my 
> Linksys
> 
> Paul
> 
> - Original Message - 
> From: "Marcel van Kaam, Fonetica" <[EMAIL PROTECTED]>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> 
> Sent: Wednesday, June 29, 2005 12:28 AM
> Subject: RE: [Asterisk-Users] Linksys WRT54GP2-NA settings 
> forperformanceandlow bandwidth?
> 
> 
> > You can set, in the linksys, the codec G729 for your line. In the Linksys
> > also set only to use that codec. This can be done at the admin page of the
> > line you use in the linksys. Also do that in the asterisk for your device.
> > First buy the license from Digium.
> >
> > Then you will use less bandwidth and have a better sound upstream.
> >
> > Marcel
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Paul 
> > Fielding
> > Sent: woensdag 29 juni 2005 1:24
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for
> > performanceandlow bandwidth?
> >
> > Hmm... Except that if I bring my Vonage ATA for my Vonage line with me to
> > the same hotel, I can get reliable connectivity.   Assuming the hotel 
> > isn't
> > helping me on the QOS front, and the Hotel's connectivity is the last 
> > word,
> > then my Vonage ATA should be choppy, as well, no?  This is what leads me 
> > to
> > think I can do some tweaking
> >
> > later,
> >
> > Paul
> > - Original Message - 
> > From: "Greg Oliver" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Tuesday, June 28, 2005 2:17 PM
> > Subject: Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for
> > performanceand low bandwidth?
> >
> >
> >> Nothing you can do on this one..  Without the provider accepting your
> >> QoS settings, you are at their mercy.  And yes, you are correct, most
> >> multi-tenant dwellings use xDSL for their connectivity due to it's
> >> price, and the upstream is usually less bandwidth than the downstream..
> >>
> >> -Greg
> >>
> >> On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:
> >>> So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
> >>> a phone at my hotel rooms, etc.   During the day or late at night the
> >>> thing works great - best ATA I've ever used.
> >>>
> >>> However, in the mid-evening (when many business travellers are at the
> >>> hotel room doing work), the outgoing audio channel gets so choppy that
> >>> the person on the other end can't make me out clearly.
> >>> Interestingly, I can usually hear them just fine - I attribute that to
> >>> larger incoming bandwidth than outgoing on the hotel's part.
> >>>
> >>> This device has a *lot* of settings that one can tweak.   Anyone have
> >>> any suggestions on tuning this thing (or tuning Asterisk or both) to
> >>> improve the SIP performance of the audio from the Linksys to the
> >>> server to try to reduce choppiness?   I note that Vonage, who also
> >>> uses these devices, seems to have got it down - it doesn't seem to
> >>&g

Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration

[Greg Oliver]

http://www.gnugk.org/compiling-gnugk.html

Also, the reqs for the included 323 channel and gnugk differ on
versions.  I have unreliably gotten them both to run on the same box
with 100% reliability.  Outbound calls transcoded from SIP -> 323 ->
Gnugk -> CCM -> MGCP -> PRI get dropped from DRQ after 2-4 seconds..

The README in the included channels/h323/README file will give you
versions for openh323 and owlib that do not match any known working
gnugk combo.  Plus some applied patches from Janus.

There is the new ooh323 channel driver out too (look on voip-info.org
for info).  I have not tried it as of yet, but it does not require
openh323 and pwlib..  That combo and gnugk on same box may work well??
It is relatively new though.

-Greg

On Wed, 2005-06-29 at 14:28 +0100, Barney Sowood wrote:
> On Sat, Jun 25, 2005 at 07:58:24PM -0500, Greg Oliver wrote:
> > That works well.  You may also want to make sure your compatibility
> > matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities
> > cause more issues than I care to talk about.  The GNUGk web site has the
> > best matrix to follow..
> 
> Do you have a specific URL, the only thing I can find is
> http://www.gnugk.org/interoperability.html, which doesn't sound
> exactly like what you're talking about.
> 
> Thanks,
> 
> Barney.
> 

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Re: [Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?

[Greg Oliver]

Nothing you can do on this one..  Without the provider accepting your
QoS settings, you are at their mercy.  And yes, you are correct, most
multi-tenant dwellings use xDSL for their connectivity due to it's
price, and the upstream is usually less bandwidth than the downstream..

-Greg

On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:
> So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
> a phone at my hotel rooms, etc.   During the day or late at night the
> thing works great - best ATA I've ever used.
>  
> However, in the mid-evening (when many business travellers are at the
> hotel room doing work), the outgoing audio channel gets so choppy that
> the person on the other end can't make me out clearly.
> Interestingly, I can usually hear them just fine - I attribute that to
> larger incoming bandwidth than outgoing on the hotel's part.
>  
> This device has a *lot* of settings that one can tweak.   Anyone have
> any suggestions on tuning this thing (or tuning Asterisk or both) to
> improve the SIP performance of the audio from the Linksys to the
> server to try to reduce choppiness?   I note that Vonage, who also
> uses these devices, seems to have got it down - it doesn't seem to
> matter where I use my Vonage Linksys device, I can get pretty
> reasonable performance.   So I figure I should be able to do similar
> tweaks to mine... *shrug*
>  
> regards,
>  
> Paul
>  
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Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration

[Greg Oliver]

We have successfully connect * .9x && 1.0.x with CCM 3.3.x and up using
both gatekeeper and no gatekeeper..  Using SIP usually with CCM 4.0 and
up..  With CCM 3.3.x, there is a limitation where the gateway H323 in
your case cannot use IP addresses, so the Asterisk box has to have
correct DNS entries to resolbve your asterisk ox..  Then just use
regular route patterns and direct it to asterisk..

That works well.  You may also want to make sure your compatibility
matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities
cause more issues than I care to talk about.  The GNUGk web site has the
best matrix to follow..

Thanks,

GReg



On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote:
> Use a gatekeeper and have both boxes register with the gatekeeper.  That 
> way you can specify what numbers go where.  From everything I have 
> tested, * will NOT register with CCM.  When I added in a gatekeeper and 
> had both sides register with it, everything works.
> 
> Walid Azab wrote:
> > Hello,
> >  
> > I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED]  > PROTECTED]> 
> > latest version. I have earlier tried getting Asterisk to register with 
> > CCM via H323 and failed. Back then, I learned that this is a known bug 
> > in Asterisk. Also people who tried doing that had also succeeded in 
> > getting calls to go through only one direction like from CCM to 
> > Asterisk. I am not that expert so excuse my ignorance with this subject. 
> > So please if anyone has any useful information or is sure that this can 
> > now work please send me whatever you have on that.
> >  
> > I simply want Asterisk users to get their dial tones through CCM.
> >  
> > Thanks and I appreciate your assistance.
> >  
> > Walid
> >  
> >  
> > 
> > 
> > 
> > 
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RE: [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems

[Greg Oliver]

OS79XX is not used in > 6 versions of firmware..  You can also, safely
go straight to 7.4.

Thanks,

Greg

On Fri, 2005-06-24 at 08:13 -0400, Tom Rymes wrote:
> For what it is worth, this is not what I did. Since you have already
> upgraded two phones to 7.4, I will assume that you know how to do that
> properly. Actually, a quick run-down:
> 
> 1.) install SIP v5.x by specifying the image name in the OS79XX.txt file
> and the SIP.cnf file. The file name should be the same in both
> files.
> 2.) Upgrade to SIP v7.0 (This step might be unneccessary, but I'm not
> sure). You need to put "P003-07-0-0" in the OS79XX.txt file, and
> "P0S3-07-0-0" in the SIP.cnf file. Make sure that you have the
> P0S3-07-0-00.loads file in the tftpboot directory, too!
> 3.) Now load the v7.4 firmware by specifying "P003-07-4-0" in the
> OS79XX.txt file, and "P0S3-07-4-0" in the SIP.cnf file. Make sure
> that you have the P0S3-07-4-00.loads file in the tftpboot directory,
> too!
> 
> One note to add, make sure that you copy the .zip file to the asterisk
> box first, thenunzip it. Do not unzip it on your PC and copy the files
> individually. Also double check permissions on the files (though again,
> you have had no problems with the othe rtwo phones.
> 
> If you can't get to the settings portion of the phone to manually
> specify the TFTP server and the Network address, I would recommend that
> you try downgrading the firmware. It's possible that the phone has the
> correct settings, but the upgrade is failing. 
> 
> Failing all else, you could install a DHCP server that specifies the
> TFTP server to the phone so that it gets the right address from the DHCP
> server.
> 
> Tom
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Tarpo, Louie
> > Sent: Thursday, June 23, 2005 7:29 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Re: Cisco 7960 firmware upgrade 
> > promblems
> > 
> > 
> > If you've followed the advice of previous posters, delete 
> > your 7-4 firmware and re-extract it.  You should NOT rename it.
> > 
> > Your OS79XX.txt should contain
> > P003-07-3-00
> > 
> > and your SIPMAC.cnf should contain
> > # SIP Configuration Generic File (start)
> > image_version: P0S3-07-4-00 
> > 
> > The P003-07-3-00 is the loader file
> > the loader file loads the actual image which is the P0S3-07-4-00
> > 
> > Louie
> > 
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of 
> > Patrick Lidstone (Personal E-mail)
> > Sent: Thursday, June 23, 2005 3:30 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems
> > 
> > 
> > 
> > > Make sure that you have done the following:
> > >
> > > 1.) Set up the phone to use DHCP to get an address *or* manually
> > > configured an e-mail address using the settings on the phone.
> > >
> > > 2.) Set the DHCP server to give out the correct TFTP server address,
> > > *or* configure "Alternate TFTP Server" = yes and manually 
> > specify the  
> > > server address.
> > >
> > > I assume that you have already done that, but you never know!
> > >
> > > Tom
> > 
> > Hi Tom, the problem is that the Universal Application Loader 
> > has never successfully loaded a firmware image, so there is 
> > no way to set these options manually. The phone is definitely 
> > doing something with DHCP, but never generates a TFTP request 
> > - apparently?
> > 
> > Patrick
> > 
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> 
> 
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Re: [Asterisk-Users] Cisco 7960 mic generating noise on other end

[Greg Oliver]

I have had several issues flashing between SCCP/SIP/MGCP on those phones
where it will eventually cause the handset to bleed through the
speakerphone.  Once that happens, the phone is basically trash - it
never stops...

-Greg



> > I'm having a problem with one of our 7960.  They all run latest 7.4
> > SIP firmware.
> > 
> > The problem appears on the other end.  The other end constantly hears
> > a 'crackling' noise.  I have tested using phone set, headset and
> > speaker and the noise appears on all cases.  I have other 7960 setup
> > exactly same way (using same asterisk, firmware, etc) so it looks like
> > a hardware issue.
> > 
> > I'd appreciate if anyone has any insight on this or any other similar
> > issues before I open the thing.


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Re: [Asterisk-Users] How to handle one incoming call on multiple lines?

[Greg Oliver]

We use it like this if that is what you are looking for:

exten => s,4,GotoIfTime(8:30-17:00|mon-fri|*|*?open,s,1)

-Greg

On Wed, 2005-06-08 at 11:24 -0400, Henry Coleman wrote:
> This feature is called  "attendant - night answer position". Is it not 
> possible to switch the incoming call to an alternate extension based on 
> time of day ?
> 
> Henry
> 
> Florian Overkamp wrote:
> 
> >Hi, 
> >
> >  
> >
> >>-Original Message-
> >>Thanks, but it isn't an option because the Telco is actually 
> >>connected to
> >>a PBX which is connected to Asterisk which should act as a intelligent
> >>answering device during non-office hours. The PBX isn't 
> >>capable of doing
> >>this. Any other option?
> >>
> >>
> >
> >Hmm, this is a bit of a hack, but it might suit your needs:
> >
> >- Make sure each of those lines goes into a different extension or context
> >- Add a delay on each line, like this:
> >
> >exten = line1,1,Do stuff
> >
> >exten = line2,1,Wait(2)
> >exten = line2,1,Do stuff
> >
> >exten = line3,1,Wait(4)
> >exten = line3,1,Do stuff
> >
> >exten = line4,1,Wait(6)
> >exten = line4,1,Do stuff
> >
> >Could this help your case ?
> >
> >Florian
> >
> >
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> >  
> >
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Re: [Asterisk-Users] Features.conf - atxfer

[Greg Oliver]

You can use super-valet-parking

On Tue, 2005-06-07 at 06:18 -0500, Mike Holloway wrote:
> Reading through the code, I don't see a way of exiting the transfer and 
> regaining the call with the customer, unless the third party hangs up or 
> maybe doesn't answer and the dialplan doesn't do anything else with the 
> call (send the call into voicemail).
> 
> I suggest you request this feature (http://bugs.digium.com), but as an 
> interim solution you can create a dialplan for internal extensions that 
> does not send the call to voicemail if unanswered, and only dials the 
> third party for a limited amount of time (20 seconds?).
> 
> You could preface these special extensions with a sequence, such as 9, 
> or 777 or whatever. Assuming your extensions are 1xx:
> 
> exten => _7771XX,1,Dial(SIP/${EXTEN:3},20)
> exten => _7771XX,2,Hangup
> 
> -mike
> 
> 
> Mark Johnson wrote:
> > I am trying out the new atxfer feature from CVS-HEAD.  I set atxfer 
> > equal to *7 and it seems to work OK.  I am having a problem getting it 
> > to work the way a receptionist would want.  If an extension calls me, I 
> > hit *7 and I hear the voice say "transfer".  I dial another extension.  
> > If the newly dialed extension goes to voicemail, I can't figure out how 
> > to get the original call back to tell them the person they are trying to 
> > reach is unavailable.  Anything I try bridges the call and the caller go 
> > into like the 2nd half of the voicemail greeting.  Is there some trick 
> > to this?
> > 
> > Thanks!
> > 
> > Mark
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Re: [Asterisk-Users] Transfer differences between BudgeTone101 and Snom190

[Greg Oliver]

You can try the ${RDNIS} variable.


On Tue, 2005-06-07 at 00:32 +0200, Elwin Andriol wrote:
> Hello all,
> 
> This email is intended rather informative than questioning. While 
> developing some script-generated dial plan, we figured out that there 
> are differences between Snom 190's and BudgeTone 101's relating to 
> transfers.
> 
> It appeared that the 190's will have their own 'Caller ID' set as the 
> 'CALLERID' variable in astersisk when transfering a call, while the 
> 101's will have the initial caller's ID set as the 'CALLERID' variable 
> on transfering a call.
> 
> We do some 'line access' determination based on an internal SIP device's 
> caller ID. It worked fine for 190's (both for normal calls and on 
> transfers) but it failed on the 101's on transfers, because the latter 
> doesn't give it's own caller ID (or at least it won't get loader into 
> the asterisk's CALLERID variable). I've found some references that said 
> that the differences between the 190's and 101's might be the result of 
> unclear definitions the the SIP transfer RFC.
> 
> Anyone figured out how I might get a transfering party's caller ID on a 
> BT101? Otherwise I will have to trash the 101's (I hate these garbage 
> phones anyway, but it would be nice for customers if they're able to 
> choose between 101's and more expensive 190's).
> 
> Using asterisk 1.0.7 + bristuff RC8g; version 3.56m on the snom190's and 
> version 1.0.6.3 on the BT101's.
> 
> regards, Elwin Andriol
> 
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Re: [Asterisk-Users] Outgoing spool file ignored

[Greg Oliver]

On Tue, 2005-05-17 at 09:26 +0100, tim panton wrote:
> On 16 May 2005, at 22:54, Jean-Denis Girard wrote:
> 
> > Andres Paglayan a écrit :
> >
> >> File::copy does copy, it re-writes the file,
> >> you need to move it.
> >> so when the the pointer is placed the file is already there.
> >>
> >
> > Well from File::Copy man page, about the move() function:
> > "If possible, move() will simply rename the file."
> 
> > I thought it was the case on Linux, but I'll probably change to  
> > system(mv ...) just to be sure.
> >
> 
> The 'if possible' thing relates to filesystem design.
> Almost all of the native UNIX filesystems support mv as an atomic action
> - but only within the same filesystem.
> (Imagine you create the file on one physical disk then 'move' it
> onto a different disk - the kernel has no option but to
> copy the file).
> 
> So create your file in a temp directory on the _same_ file system as
> the destination, then do the move.
> 
> If your filesystem is remote (samba or nfs) or non unix native (FAT)
> then it just won't work.
> 
> Tim.
> 
> 
> > Thanks for all the replies.
> > Jean-Denis
> >
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Re: [Asterisk-Users] Call Manager Express Peer

[Greg Oliver]

The only thing I have different in my CME dial-peers is "application 
session" for each of them.  Other than that, what you have works for me..

-Greg
Nathan Alberti wrote:
I have the following configuration and am obviously missing something 
small that is causing * not to work as expected.

I have the following defined in sip.conf
[ccme-in]
type=peer
host=10.0.9.1
context=devel_in
disallow=all
allow=alaw
nat=no
canreinvite=yes
qualify=yes
and [devel_in] is defined in extentions.conf
However when I try to call via the dial peer I have configured on the 
cisco (below) I get :

Feb 22 18:37:40 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'

Which is correct, meaning the context declaration is not being respected.
--
dial-peer voice 101 voip
destination-pattern 10.
session protocol sipv2
session target ipv4:10.0.0.133
dtmf-relay rtp-nte
codec g711ulaw
no vad
---
My bad or something else ??
TIA,
Nathan.

Here is a sip debug for that peer:
Sending to 10.0.9.1 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.0.9.1:19206
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 101 in default
Feb 22 18:44:25 NOTICE[31486]: pbx.c:1318 pbx_extension_helper: Cannot 
find extension context 'default'
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.9.1:5060;branch=z9hG4bK3047A
From: "Test Phone 1" ;tag=17AFD44-10AD
To: ;tag=as3edc130d
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

to 10.0.9.1:5060
Destroying call '[EMAIL PROTECTED]'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.9.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.133:5060;branch=z9hG4bK2b06e290
From: "asterisk" ;tag=as0a8b5343
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Feb 2005 10:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.9.1:5060
Destroying call '[EMAIL PROTECTED]'

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Re: [Asterisk-Users] Multiple Parking Lots.

[Greg Oliver]

We use it for a client in 2 ways...
exten => _218X,1,SuperValetParking($[ ${EXTEN} + 100 ]|mylot|15|$ [${EXTEN} + 
100]|10|superpark)
exten => _218X,10,Playback(vm-nobodyavail)
exten => _218X,11,Dial(SIP/${OPERATOR},15,m)
exten => _218X,12,Hangup

exten => _228X,1,SuperValetParking(${EXTEN}|mylot|15|${EXTEN}|10|superpark)
exten => _228X,10,Playback(vm-nobodyavail)
exten => _228X,11,Dial(SIP/${OPERATOR},15,m)
exten => _228X,12,Hangup
This lets them park to 218x and pick at 228x
exten => 2181,1,System(cp /var/spool/asterisk/scripts/2191 /var/spool/asterisk/outgoing/)
exten => 2181,2,SuperValetParkCall(2191|mylot|25|2181|7|superpark)
exten => 2181,7,NoOp(Hangup SideCar Call)
exten => 2181,8,SoftHangup(SIP/12191)
exten => 2181,9,System(rm -f /var/spool/asterisk/outgoing/2191)
exten => 2181,10,Playback(vm-nobodyavail)
exten => 2181,11,Dial(SIP/${OPERATOR},45,m)
exten => 2181,12,Hangup 
This example lights the lamp on a 7914 sidecar to show there is a call 
parked at that extension..

You can just put the parking spots in different contexts to keep them 
separate from different groups of users..


Chris Modesitt wrote:
Thank you any feedback would be greatly appreciated.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, February 22, 2005 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Parking Lots.
juse valetparking. I can't give you an example right now, since I'm
working on implementing it. When I'm done I'll report
On Tue, 22 Feb 2005 11:51:45 -0700, Chris Modesitt <[EMAIL PROTECTED]>
wrote:
Question: I am PBX multi-hosting several customers on one of my * servers,
what the best way to setup call parking to prevent company A from picking
up
Company B's parked calls ?

Any basic examples would be greatly appreciated.

Thanks

Chris.
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Re: [Asterisk-Users] Asterisk "no one is available to take your call"

[Greg Oliver]

True, but it also states that with no timeout value that it will dial 
until the caller hangs up.

I have included my dial pattern - can anyone see anything that would 
cause this, or something in my sip.conf or h323.conf files that would 
override these settings?

Thanks,
Greg Oliver
[outbound]
exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} )
exten => _4XXX,2,Dial(H323/${EXTEN})
exten => _4XXX,3,Congestion
exten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten => 
_5XXX,2,Dial(H323/${EXTEN})
exten => _5XXX,3,Congestion

exten => _9NX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _9NX,2,Dial(H323/${EXTEN})
exten => _91NX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _91NX,2,Dial(H323/${EXTEN})
default context includes outbound, and contexts in sip.conf and 
h323.conf are using default.  Like I say, call answered before ~5 
seconds are fine, other than that it is transferred to 4607..

Howard Lowndes wrote:
On Wed, 2005-02-16 at 11:05, Greg Oliver wrote:
OK - I can successfully make calls from SIp phone through an asterisk 
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.

The problem is that if the call is not answered within ~5 seconds, * 
gives the message "no one is available to take your call" and 
disconnects the call.  If I answer b4 the 5 seconds - everything is good.

Anywhere I need to set to get around this.
I have tried the t,T settings (even though the docs say no entry is 
forever) with no luck.

Read the doco on the Dial command again.  It's noting to do with the Tt
option, it's the parameter before that that you need to set to the
timeout
Thanks,
Greg Oliver
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[Asterisk-Users] Outbound calling timeout

[Greg Oliver]

I am running asterisk 1.0.1 with OH323 compiled in.
We have a 323 trunk to CallManager with a mgcp controlled pri router.
When using sip phones (directly registered with asterisk) to call out 
the 323 trunnk to PSTN, calls timeout after 3 rings.  If I answer b4 3 
rings - no problem, otherwise I get "no one is available to answer at 
this time" on the consoel and it redirects to an extension in 
extensions.conf under a different context.

Any ideas on where I should be looking:
Thanks,
Greg Oliver
configs follow:
sip.conf
sip*CLI>
sip*CLI>
sip*CLI> exit
Executing last minute cleanups
[EMAIL PROTECTED] asterisk]# cat sip.conf
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
;   sip show peers  Show all SIP peers (including friends)
;   sip show users  Show all SIP users (including friends)
;   sip show registry   Show status of hosts we register with
;
;   sip debug   Show all SIP messages
;
[general]
context=default ; Default context for incoming calls
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to 
RFC 3261
; Set this to your host name or domain name
port=5060   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=no; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
;pedantic=yes   ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility
;tos=184; Set IP QoS to either a keyword or numeric val
;tos=reliability  ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing 
registration
notifymimetype=text/plain   ; Allow overriding of mime type in NOTIFY
;videosupport=yes   ; Turn on support for SIP video
disallow=all; First disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
;allow=ilbc ; Note: codec order is respected only in 
[general]
;musicclass=default ; Sets the default music on hold class for all 
SIP calls
; This may also be set for individual 
users/peers
;language=en; Default language setting for all users/peers
; This may also be set for individual 
users/peers
;relaxdtmf=yes  ; Relax dtmf handling
;rtptimeout=60  ; Terminate call if 60 seconds of no RTP 
activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP 
activity
; when we're on hold (must be > rtptimeout)
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;   register => user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a
; section defined below.
;
; Examples:
;
;register => 1234:[EMAIL PROTECTED]
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'.  Calls from this provider 
connect to local
;extension 1234 in extensions.conf default context, unless you define
;unless you configure a [sip_proxy] section below, and configure a context.
;Tip 1: Avoid assigning hostname to a sip.conf section like 
[provider.com]
;Tip 2: Use separate type

[Asterisk-Users] Asterisk "no one is available to take your call"

[Greg Oliver]

OK - I can successfully make calls from SIp phone through an asterisk 
323 channel to a Cisco Call Manager and out a MGCP controlled gateway.

The problem is that if the call is not answered within ~5 seconds, * 
gives the message "no one is available to take your call" and 
disconnects the call.  If I answer b4 the 5 seconds - everything is good.

Anywhere I need to set to get around this.
I have tried the t,T settings (even though the docs say no entry is 
forever) with no luck.

Thanks,
Greg Oliver
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[Asterisk-Users] Asterisk 1.0.1 - CCM 3.0.3 - GNUGK 2.0.8 - OpenH323

[Greg Oliver]

OK - I have an cisco MGCP gateway controlled by a Cisco Call manager.
There is a gatekeeper controlled 323 trunk between callmanager and 
asterisk (we use asterisk for vmail, sip phones, park/pick/meetme, etc..).

We are using GK routed mode in gnugk.  All extensions in h323.conf and 
callmanager register with gatekeeper just fine and can communicate.

Dialing sccp phones from sip through the trunk worked just fine, but 
dialing sip from sccp would not.  I had to add the sip extension in 
h323.conf so it would register with the gatekeeper, and then I could 
successfully call it from sccp..  Is there a better way to accomplish this?

Also, SIP phones from asterisk cannot use the mgcp gateway through call 
manager - the gatekeeper just keeps saying  not registered 
as an endpoint.  I have tried setting all of the all unregistered, etc 
settings in gatekeeper.ini to allow traffic to pass through with no luck.

Does anyone know how I can get the gatekeeper or asterisk to pass calls 
through callmanager which will in turn pass it to the mgcp gateway for 
outbound calling?

Thanks,
Greg
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Re: [Asterisk-Users] RE: Cisco 7960G phone crashes during SIP upgrade

[Greg Oliver]

The 7.3 zip file contains the wrong filenames from their website.  If 
you watch the status messages, you should see a "incorrect loads"  or 
"invalid loads" flash before the phone continually reboots.  I cannot 
remember exactly which files I renamed to make it work, but I edited the 
.loads file, and renamed sbn or bin, etc and finally got it all working 
to 7.3 from 6.5 Skinny image.

-Greg
Adi Linden wrote:
I can confirm that the Cisco instruction for installing/upgrading to the
7.3 SIP image do not work. When I originally installed SIP on some brand
new phones Cisco TAC indicated that a phone has to run 6.3 before it can
be upgraded to 7.3.
Loading the 6.3 SIP image has been a success. I've had no problems
switching between 6.3 SIP and various SCCP images back and forth.
I might try your suggested method to get up to the 7.3 SIP image.
However, it doesn't sound like a hassle free method of loading SIP at all.
I'd much prefer if I could tell the phone via the dhcp server to load from
tftp server x.x.x.x and automatically loads SIP or load from tftp server
y.y.y.y and automatically load SCCP (where x.x.x.x is Asterisk and y.y.y.y
is Cisco CCM).
Adi
On Thu, 3 Feb 2005, Gene Willingham wrote:

I found the phones to be very flaky.
Make SURE:
1.  The txt files are unix txt files. NO CR/LF at end of line.  If you edit
the files on a PC use wordpad only.  If you use Notepad they will ADD extra
characters that causes the phone to reject the files.  Do not save them as
Unicode text files, this is cause it to fail as well.
2.  The 7.x SIP image is not a full image, you have to start with 6.3 or
6.0.  I have had success starting with 6.3 on some phones, but others
required me to start with 6.0
3.  Upgrade path 6.0 > 6.3 > 7.3
4.  The 7.3 upgrade fails because the directions on how to upgrade are
wrong.  Try this.
a.  Edit OS79XX.TXT and put P003-07-3-00 as image name
b.  Edit SIPDefault.txt and put image name as P003-07-3-00
Reboot the phone.
The phone will upgrade the universal loader application, then fail
on loading the 7.3 application w/ a "Protocol Application Invalid" error.
DO NOT UNPLUG THE PHONE:
c.  Now Edit SIPDefault.txt and change image name to P0S3-07-3-00
When the phone reboots by itself, it will upgrade the SIP image to
7.3  and hopefully you are done.  I have found with no explanation that
sometimes the phone will take multiple reboot with errors before it will
work.  Cisco says there is a "Checklist" of things it is looking for.
Apparently this checklist progresses from each successive reboot.  If you
unplug the phone all you are doing is starting over again.  Be patient it
could take 20 - 30 minutes for each phone.
I am not an expert, I have successfully upgraded 20 phones.  This was the
best I could figure out through trial and error.
If you get a checksum error on the SIP 6.3 image, the problem is with your
config files not the image.
Good luck.  I have found upgrading polycoms easier.  Found the phone quality
to be very good.  They have a bug in their 1.3 image that affects phone to
phone dialing on the polycom.  That appears to have gone away when I
upgraded to 1.4
Gene
--
Message: 8
Date: Thu, 3 Feb 2005 13:18:28 +
From: Nicolas Chabbey <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=US-ASCII
Hello,
I've recently received a Cisco 7960G phone with the factory default
SCCP firmware on it.
As we're using SIP on our network, the first things i've done was to
upgrade but unfortunately the phone just restarted. By looking on the
TFTP logs and tcpump output, i've seen that the phone crashed and
restarted just after downloading the OS79XX.TXT file, without
requesting the image file at any moment.
If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT
(begining with P003), the phone doesn't crash and request the
respective SEP.xml file. Unfortunately (again), just after
downloading the xml configuration it hang and restart. I've checked
the syntax and they's no error on it, if they's one the phone output
the error on the display without crashing. Note that i've both put
with and without the load information statement, with the same result.
Both statical and DHCP configuration has been tried.
Maybe it's an hardware failure or i've miss somethings realy important :)
Thanks

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Re: [Asterisk-Users] reason 24 (Call ended with Q.931 cause)

[Greg Oliver]

Turn on "debug isdn q931" "term mon" on your 5350.  It is an ISDN 
signalling error.  Strange it is showing up in asterisk through a 323 
trunk though...

What happens when you do a "csim start xxx" where xx = phone 
number to dial from the 5300?

-Greg
Tola Ogunsan wrote:
Hi Michael  and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm 
getting this error

"reason 24 (Call ended with Q.931 cause)"
I've checked the Asterisk wiki and several other resources.  Please can 
anyone give me a hint on what the problem is I reach my wits end.  Thanks

Tola
my config and debug
Configuration of OpenH323 channel driver
--
Version: 0.7.1
Listening on address: 0.0.0.0:1720
Gatekeeper used:  No gatekeeper
FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF
Supported formats in pref. order: g729<0>
Jitter buffer limits (min/max): 20-500 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: 3
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 10
Max call rate (ingress direction): 1.00/30
Starting simple switch on 'Zap/3-1'
  -- Executing Wait("Zap/3-1", "1") in new stack
  -- Executing Dial("Zap/3-1", "OH323/[EMAIL PROTECTED]|10") in 
new stack
  -- H.323 call to [EMAIL PROTECTED] with codec(s) g729
Outbound H.323 call 'ip$localhost/263'.
  -- Called [EMAIL PROTECTED]
Call 'ip$localhost/263' cleared.
  -- H.323 call 'ip$localhost/263' cleared, reason 24 (Call ended with 
Q.931 cause)
Call 'ip$localhost/263' cleared in INIT state.
  -- OH323/L263 is busy
  -- Hungup 'OH323/L263'
== Everyone is busy/congested at this time (1:1/0/0)
  -- Executing Hangup("Zap/3-1", "") in new stack
== Spawn extension (incoming, s, 3) exited non-zero on 'Zap/3-1'
  -- Hungup 'Zap/3-1'
Call 'ip$localhost/263' without owner has already been cleared (2).
  -- Starting simple switch on 'Zap/3-1'

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[Asterisk-Users] 7900 Problem with Asterisk 1.0.1 and OH323

[Greg Oliver]

OK - I have posted this question before, but no responses.
I have built 3 * servers since and can successfully connect to all 3 as 
well as another providers with my 7960.  We have a server at work that 
uses OH323 to interface with Cisco CallMangler, and for the life of me I 
cannot get my 7960 to authenticate with it.  X-Lite authenticates and 
registers, but the 7960 just gives 401 UnAuth errors on the console.

Any ideas on why this would happen.  Are there any compile options that 
could be set to cause this?  md5 auth with x-lite works fine - I have 
tried md5 and plaintext with 7960, but never authenticates..  The 
console of the 7960 shows provisioned, but not authenticated or 
registered.  I am running the 7960 behind NAT, but have been doing so 
for years and it works to 2 * servers on the internet and 1 inside the 
NAT simultaneously, so I know I have my NAT settings right..

Any Ideas?
Thanks,
Greg Oliver
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Re: [Asterisk-Users] Configuring VLAN takes ages

[Greg Oliver]

I am using 7.3, and do not experience this behavior on Linksys or Cisco 
powered switches not configured for voice vlan - I have a pre G model of 
the phone though.  I just cannot get mine to authenticate :(

-Greg
Mark Johnson wrote:
Asterisk wrote:
when booting the cisco 7960 with SIP image 7.3, the "Configuring VLAN" 
takes in order of minutes before it issues a DHCP request .

Does anyone else have this problem - is there any way of disabling the 
VLAN configuration at all ?

We are not using Cisco switches.
Julian

I upgraded to 7.3 yesterday and am having the same problem using Cisco 
switches.
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[Asterisk-Users] Asterisk v1.0.1 Cisco 7960 Sip v7.3

[Greg Oliver]

Running those versions of code, my 7960 will not register with Asterisk.
The same 7960 is authenticating against another * server on line 2 just 
fine though - with the same settings in sip.conf.  On the failing * 
server I am just getting 401 unauthorized errors on the console.  From 
the phone's shell I get that t is registering, but not authenticated .1. 
from show reg.

Any ideas would be appreciated.  Only passwords were removed.
Thanks,
Greg Oliver
I have included my SIP.cnf file for review..
# SIP Configuration Generic File (start)
# Proxy Server
proxy1_address: "sip.cistera.com"
proxy2_address: "pbx-nwcorp.nationwide.net"
proxy3_address: "192.168.117.4"
proxy4_address: "192.168.117.4"
proxy5_address: "192.168.117.4"
proxy6_address: "192.168.117.4"
# Line 1 Settings
line1_name: "74678"  ; Line 1 Extension\User ID
line1_shortname: "Cistera"
line1_displayname: "Cistera"   ; Line 1 Display Name
line1_authname: "74678" ; Line 1 Registration Authentication
line1_password: "" ; Line 1 Registration Password
# Line 2 Settings
line2_name: "131"  ; Line 3 Extension\User ID
line2_shortname: "MobilPro" ; Line 3 Extension\User ID
line2_displayname: "Sarah"   ; Line 3 Display Name
line2_authname: "131" ; Line 2 Registration Authentication
line2_password: "" ; Line 2 Registration Password
# Line 3 Settings
line3_name: "" ; Line 2 Extension\User ID
line3_shortname: ""
line3_displayname: ""   ; Line 2 Display Name
line3_authname: "UNPROVISIONED" ; Line 3 Registration Authentication
line3_password: "UNPROVISIONED" ; Line 3 Registration Password
# Line 4 Settings
line4_name: ""  ; Line 4 Extension\User ID
line4_displayname: ""   ; Line 4 Display Name
line4_authname: "UNPROVISIONED" ; Line 4 Registration Authentication
line4_password: "UNPROVISIONED" ; Line 4 Registration Password
# Line 5 Settings
line5_name: ""  ; Line 5 Extension\User ID
line5_displayname: ""   ; Line 5 Display Name
line5_authname: "UNPROVISIONED" ; Line 5 Registration Authentication
line5_password: "UNPROVISIONED" ; Line 5 Registration Password
# Line 6 Settings
line6_name: ""  ; Line 6 Extension\User ID
line6_displayname: ""   ; Line 6 Display Name
line6_authname: "UNPROVISIONED" ; Line 6 Registration Authentication
line6_password: "UNPROVISIONED" ; Line 6 Registration Password
# Emergency Proxy info
proxy_emergency: ""
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: ""
# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "64.123.190.68"
voip_control_port: "5060"
start_media_port: "16000"
end_media_port:  "32768"
nat_received_processing: "1"
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Home  "; Has no effect on SIP messaging
# Time Zone phone will reside in
time_zone: CST
# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2"  ; 0-Disabled (default), 1-Enabled, 2-Privileged
# Phone prompt/password for telnet/console session
phone_prompt: "Get Out"  ; Telnet/Console Prompt
phone_password: "a"  ; Telnet/Console Password
# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"
# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: "none"
# URL for external Directory location
#logo_url: "http://10.0.1.3/10-20logo.bmp";; URL for 
branding logo to be used on phone display

# SIP Configuration Generic File (stop)

SIP.CONF entry:
[74678]
context=default
type=friend
username=74678
secret=
host=dynamic
reinvite=no
canreinvite=no
nat=yes
disallow=all
allow=ulaw
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