[Asterisk-Users] Cisco 2600 Gatekeeper registrations
Hello. I am trying to connect Asterisk to Cisco Gatekeeper running on 2600 router. On the Gatekeeper I get the loopback address from Asterisk in "CallSignalAddr". All Other devices that registers on this Gatekeeper report their ethernet ip address in this section. router1#show gatekeeper endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 127.0.0.1 1720 10.169.208.12 32774 router1 VOIP-GW H323-ID: mailbox E164-ID: 4557999 H323-ID: det-gw H323-ID: time E164-ID: 155 Voice Capacity Max.= Avail.= Current.= 0 in /etc/asterisk/h323.conf are the following lines: [general]port = 1720bindaddr = 10.169.208.12 gatekeeper = 10.169.208.1AllowGKRouted = yes context=default [time]type=h323e164=155context=default [det-gw]type=h323prefix=1248,1313context=detroit [mailbox]type=h323e164=4557999context=default ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem facing on Firewall, NAT and asterisk
Hi Prasad, Install a Asterisk in your DMZ and one Asterisk inside of your Lan. Set them to use IAX between them passing through your firewall. A) Your SIP Phones in your lan will connect to your LAN's *. B) The SIP Phones in the internet will connect to your DMZ's *. C) A connnects to B through the Asterisk's IAX connection SIP doesn't work with firewalls. Also, next time, post this kind of message in the asterisk-users list. Isamar On Wed, 3 Nov 2004, prasad_s wrote: Hi all, I am using asterisk, which is running on one machine having static(global) IP. I have another machine(Internet server with global IP, with firewall) working as gateway for internal machines having local IP starting with 192.168.xxx.xxx. My SIP client(xten-xlite) is on LAN machine and registers to the asterisk server through this sip phone. All machines on the LAN, having sip phone are registered to asterisk server. But the problem is when I call internally between two sip client I don't get voice path between these two sip phones, i.e. I can not talk and hear from both phones, though I get message on the asterisk server connected. Is this because of Firewall and NAT between my sip client and asterisk server? But then how I get register to asterisk server? Is there any workaround for this problem regards Prasad Somwanshi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Console Error message
I got a strange error message on the CLI, saying: Warning: File.c:550 ast_readaudio_callback: Failed to write frame any ideas ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: FXO module in TDM400P (UK, BT) - Hangup detection failing
[apologies if too much context snipped - I'm subscribed to receive digests. correct me off-list if it's an issue] * Ian D. Willoughby (Wed, 3 Nov 2004 10:49:44 +) Are you based in Hastings by any chance (Senlac and all that)? Heh. No, Senlac is my street name in Romsey, Hampshire. I have similar problems with the voicemail but not with users dialing in with extensions ringing. I do have a problem that zap show channel shows the Zap line as OffHook after a call and it never goes back to OnHook and asterisk gets confused about channel status. Hmm. I haven't observed that last issue with mine. When the voicemail detects the silence, it hangs the FXO channel up and the local status reports as OnHook. If I understand some of the previous posts in the archive and what my multimeter tells me correctly, then the pop/click i hear is the exchange reversing the polarity of my line. The x100p obviously could detect that and correctly signal the driver that the remote party has hung up, the FXO module clearly cannot :-( Mark/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FXO module in TDM400P (UK, BT) - Hangup detection failing
On Wed, 3 Nov 2004 13:10:57 +, [EMAIL PROTECTED] wrote: On Tue, Nov 02, 2004 at 10:58:39PM +, StrUK wrote: snip other information I guess my question is: does anyone have polarity reversal hangup detection working on a BT line with an fxo module in a TDM400P? Testing with my fxo module shows that it takes about 8 seconds from pressing hangup on the mobile phone before hangup is detected in asterisk (maybe this relates to your 9 seconds). I'm not sure if this is when the local exchange sends the polarity reversal or if asterisk is detecting a hangup in some other way. I have busydetect=no and callprogress=no so I'm (AFAIK) not using any fake disconnect functions as per the keypad lights page[1]. If I pick up the phone during the 8 seconds I get a usable dialtone, which might be worrying if I didn't want people calling certain numbers. This seems to indicate that the local exchange knows that the mobile has hungup, as it wouldn't give dialtone in the middle of call. So perhaps I dont have polarity reversal hangup detection working, but HTH anyway. I'm not using the fxo callerid and asterisk is from cvs checkout date 14-Aug-2004 (as downloaded from bri-stuff.0.1.0-RC4a) my zapata config for the fxo module: snip I have UK callerid enabled and I have the same problem, just haven't got around to asking anyone about it yet! A slight variant on the usual setup is that I've got Home Highway from which I'm taking the two analog channels. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Reject a call if no callerID
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Wednesday, November 03, 2004 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, Nov 03, 2004 at 04:45:03PM +0900, Hermann Wecke said:
I couldn't think any recipe to reject a call if no callerID is presented.
PrivacyManager and Zapateller are not an option, as the call will be
answered before I can drop it. I just want to silent drop the call: no
callerID, no answer.
See example 3 in:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
Here is what I do:
exten = s,1,GotoIf($[${CALLERIDNUM} = ]?s|5)
exten = s,2,yadayada
exten = s,3,yadayada
exten = s,4,yadayada
exten = s,5,Hangup
;
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RE: [Asterisk-Users] Reject a call if no callerID
But now that logic works. However how would you insert that into the
dialplan to get it to work or would AGI be better solution?
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Wednesday, November 03, 2004 3:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, 2004-11-03 at 18:45, Hermann Wecke wrote:
I couldn't think any recipe to reject a call if no callerID is
presented.
PrivacyManager and Zapateller are not an option, as the call will be
answered before I can drop it. I just want to silent drop the call:
no
callerID, no answer.
Any ideas?
I would imagine a simple gotoif followed by hangup would suffice
in psuedo code:
if (${CALLERID} == )
then
hangup
else
goto(incoming,s,1)
fi
I am not familiar with gotoif, but show application gotoif should
help.
Regards,
Adam
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Re: [Asterisk-Users] Best codec for faxes?
steve szmidt wrote: On Tuesday 02 November 2004 08:18 pm, Julio Arruda wrote: Steve Underwood wrote: 'replace' fax over G.711 by fax over G.711 over IP :-) The point being, Fax over VOIP (even using G.711), I don't believe would be as reliable as Fax over an ISDN b-channel :-) Better now ? You're totally right. The advantage of ISDN is lost, from a fax viewpoint, largly because the unwieldly Internet don't care about lagging packets etc. Point to point ISDN is a great dedicated circuit. But it ain't that no more once you connect it to the Internet. Yes and no. Across the Internet even a G.711 FAX is unpredictable. However, people often get perfectly good results on lightly loaded LANs. It still isn't perfect, as a urst of data on the LAN can still upset things, but some people get results they can live with. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New versions now available!
Hello all, The asterisk-oh323 package has been updated. From now on, there are two series of releases: - 0.6.x releases, latest is 0.6.4. These will work with Asterisk v1-0 source code. - 0.7.x and above, latest is 0.7.0. These are for CVS code of Asterisk. Also, the latest versions now use OpenH323/Pwlib Janus-patch4 libraries. 0.6.4/0.7.0 versions contain major stability fixes and some fixes for dynamic codec negotiation (although there are still same cases that do not work). Additionally, the build process supports the building of a chan_oh323.so binary that has statically linked the OpenH323/Pwlib/oh323wrap libraries (to avoid common runtime errors with conflicting versions of OpenH323/Pwlib libraries). No new features have been added. In the following versions, I'm thinking about extending the configuration of the channel driver (oh323.conf) for supporting specific configuration options per H.323 endpoint and gateway used. Something like the user/peer/friend philosophy of Asterisk config files, but closer to the needs of chan_oh323. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] An anniversary and a lament for FXOs
Michael The problem lies not with Asterisk or the cards but with the combination of voip and analogue telephony in general. I can guarantee that you will have very similar echo when you connect your Panasonic pbx to the analogue lines. In fact, your echo will most likely be worse. It is just that you do not perceive it as echo because it occurs more or less coincident with the original sound. However, the propagation delay through the systems that is introduced by Asterisk and voip (by necessity and design) is the culprit, because this adds the delay which causes the echo to become perceptable. You will never get a match good enough to eliminate echo. What you may be able to do is to get a match that is good enough to allow the built-in echo cancellers to be effective. You don't say where you hear the echo - near end or far end? A small diagram of your system posted on your web site, with an indication of who hears the echo and which echo they are hearing would help me (or countless others!) to help you work this out. I am using 2 X100P cards here in the UK connected to extension ports of a Panasonic KXT616 and we get no perceptable echo at all. Good luck! Rgds Tim Robinson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 03 November 2004 15:14 To: Asterisk Group Subject: [Asterisk-Users] An anniversary and a lament for FXOs This week marks one year since I first setup an Asterisk server in the hopes of transitioning my home office to a total VoIP system. The process has been an incredible learning experience. I've tried numerous IP hard phones, eventually settling upon the Polycom IP600 as my choice. I've also used multiple ATAs including all the Sipura products. Using Asterisk has been a challenge, a thrill and (when its working) a joy. However, the one thing that I am not satisfied with is the performance of the FXO interfaces that bring in my PSTN lines. I've tried X100p cards but found them horribly unreliable. I presently use Sipura SPA-3000s but they're only marginally better. How is it that my Panasonic 4 line SOHO phone system (KX-TG4000B) can have four stable, reliable FXOs with no echo at all in a device with a total cost of $500? It seems to me that there ought to be hardware available that behaves just as well, but bridges the PSTN to the SIP/IAX domain? I've read a lot on the list about how difficult designing FXOs can be, but that flies in the face of the fact that every small multi-line phone system has them...and without expection those behave better than the devices I've been able to try with Asterisk. The Sipura SPA-3000 has several settings to adjust for line impedance and inductive/capacitive line loadinglots of settings, but it provides nowhere near the basic performance of one of the lines on the Panasonic KSU. It's simply mind boggling. So, while I've posted with respect to FXOs previously, I must ask againwhat FXO interface device can anyone recommend from real experience? Michael P.S. - I even investigated switching my lines to ISDN to get around the need for FXOs, but SBC won't do it where I live. -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good ringing plans for small office
On Wed, 3 Nov 2004 11:11:42 -0500, Christopher TenHarmsel
[EMAIL PROTECTED] wrote:
At the place I work we're using Asterisk to run our in-office phone
system. We have about 15 employees and a total of about 5 hard phones.
Right now when asterisk receives an incoming call, it rings all 5
phones, because we don't really have a receptionist. I was wondering
if anyone has had a similar situation and might be able to suggest some
better approach to ringing the phones? I don't really know what I have
in mind, just kind of looking for what other people in similar
situations might have done.
We have a similar environment. I ring 3 phones, but I use a different
ring so the employees know it's an outside call and can answer it
accordingly. From my extensions.conf:
exten = 0,2,Dial(${USER1}r2${USER2}r2${USER3}r2,20,Tt)
In addition, if nobody answers the call, it goes to a separate
global voicemail box that is set to email a notification to a
supervisor.
John
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[Asterisk-Users] Installing X100P Asterisk - Unable to create channel of type 'Zap'
Hello list, I am trying to install a DigiumX100P into a Redhat Asterisk.Kernel seems to be OK, card OK.Zaptel Configuration seems to be OK. # ztcfg -vvChannel map:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.Asterisk works fine with IP SIP but not with X100PI get the error on AsteriskCLI channel.c:1919 ast_request: "No channel type registered for 'Zap'"and than app_dial.c:763 dial_exec "Unable to create channel of type 'Zap'".Does anyone know what might be the problem ? Thanks for any help Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How change default law for T100P
I would like to know if there is a way to change default ulaw for a T1 card. I see in the zap show channel X that is working as ulaw. How do I change it in zapata.conf or zaptel.conf to alaw. Iam interconnecting a Meridian PBX but I need to configure it as alaw. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
