Re: [asterisk-users] Avaya 4610sw IP Phone
On 24/01/2012, at 7:46 AM, Jonn Taylor wrote: This phone only works with Avaya IP Ofiice. That's the 5610SW. The 4610SW while sharing the same appearance and also working on the IP Office (with H323 firmware) was designed for the Avaya Communications Manager, and therefore there is SIP firmware available. Aamir, it should work with the SIP firmware. I've registered the 4621SW with Asterisk in the past (the big screen version), and don't remember having any difficulties but this was a few years ago. If the phone is loaded with H323 then you'll need to replace it with SIP before proceeding. -Shaun -- Shaun Ewing | sh...@shaun.net | http://shaun.net/ smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk and avaya SIP trunking
On Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote: HI Shaun and Robb, Thanks for the assistance. I was able to force the codecs and had avaya talk in the right way. Also addressed the DTMF issues. Glad to hear it. I seem to be having issues with asterisk and avaya not detecting Hang ups. i am using the Analog phones connected to the POTS ports on the IP Office. I will try connecting the avaya Analog and Avaya IP Phone to IP Office and see if that makes any difference. What does SSA show when one end has hung up? If it still shows the call as active, then a disconnect signal has gone missing. I've never experienced this problem, but then again the only thing we use the POTS ports for is faxing and this is forced to use our PRI circuits. All of our handsets including conference room phones are IP. -Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk and avaya SIP trunking
On Mon, Nov 10, 2008 at 2:28 PM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote: Hi Guys, Thanks that did help to resolve my issue. i tried the .@10.10.8.1 and it worked and i had a successful call but i have the following 2 concerns. 1. We have voice communication from avaya to asterisk now but avaya is forcing asterisk to use only codec G723. if i disable G723, it says no compatible codecs. While the calls from asterisk to avaya are being accepted as alaw Make sure you have Compression Mode in your SIP line config on the IP Office set to your desired codec. You'll run into this problem if you have it set to Automatic Select. Make sure you also reduce the number of codecs on the Asterisk side. For example, our sip.conf entry looks like: [ipo-cbr2] type=friend username=ipo-cbr2 secret=xx host=172.31.2.1 nat=never context=from-ipo-cbr2 insecure=port,invite disallow=all allow=ulaw allow=alaw canreinvite=yes qualify=no dtmfmode=auto To reduce VCM usage, also make sure your IP handsets are using the same codec. If they are, you won't use any VCM channels for a call. 2. I am having issues with DTMF. DTMF is not being recognized or being sent from avaya to asterisk. It should work. Make sure the Asterisk side has dtmfmode=auto like above. -Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with asterisk and avaya SIP trunking
On Sun, Nov 9, 2008 at 8:19 AM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote: HI Robb, I had the checked the IP Office and i see that in the SIP Line Settings an option [checkbox] that says (Use Tel URI), which is unchecked. But i still get the Tel:+ in the SIP Header (even when it is turned on or off). I believe the use tel URI is only used for inbound calls (ie: from Asterisk to Avaya). For inbound calls to work you need to leave 'use tel URI' unchecked. When you're creating the shortcode (either in the shortcode section or in ARS), you need to add @10.10.8.1 after the number, eg: .@10.10.8.1 Example at http://www.se.id.au/miscimages/avaya-sip-ars.png Let me know how you go. I've got a few Asterisk boxes talking to our IP Office very happily (and used quite a lot), so I'm quite happy to help where I can. -Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
On Sep 5, 2007 3:36 PM, Kai-Uwe Jensen [EMAIL PROTECTED] wrote: How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. Also fixed it here. I had some quite bad jitter on the first few seconds of speech with the default setting (app_swift-2.0rc1). Searched the Asterisk archives, found your message, made the change and voila! Thanks, Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handling SIP 404 event
Hello all, I am curious, does anybody know of a way to handle the SIP 404 event? (ie: is this stored in a variable somewhere, so one can handle it in the dial plan). For example, dialing an invalid number on another softswitch on the network: -- Executing Dial(SIP/sip7110-8118, SIP/[EMAIL PROTECTED]|60|r) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 404 Not Found back from 172.16.23.31 -- SIP/softswitch-791b is circuit-busy == Everyone is busy/congested at this time At the moment my dialplan logic will respond with nobody is available to take your call at the moment. goodbye, unless that extension has a mailbox. I'd like to be able to branch out and play a different message when a 404 is received. Thanks, Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] [ANNOUNCE] chan_capi-cm-0.6 released
On 9/21/05, Armin Schindler [EMAIL PROTECTED] wrote: Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Great work Armin. I'll try to get around to testing it today :-) -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] [ANNOUNCE] chan_capi-cm-0.6 released
On 9/21/05, Armin Schindler [EMAIL PROTECTED] wrote: Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Doesn't seem to work with 1.0.8: Sep 21 10:25:13 WARNING[16435]: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: get_ast_capi_MessageNumber Sep 21 10:25:13 WARNING[16435]: Loading module app_capiCD.so failed! -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Callmanager Asterisk for Voicemail revisited
Please do! Doug Here it is: http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Voicemail+Integration It needs some cosmetic work, but I think that gets the point across :-) -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Callmanager Asterisk for Voicemail revisited
On 9/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Did you find a way to turn-on voice-mail lamp on Cisco phone connected to Cisco Call Manager, when there is new voice mail in Asterisk mailbox? Yes, this is described in the wiki document I created. -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Asterisk in Spanish
On 9/19/05, Sebastian Milioto [EMAIL PROTECTED] wrote: Hi all, I've been installing [EMAIL PROTECTED] and (of course) all the answering machine (I don't sure that's the right word in english, preatendedora in spanish) speech is in enlgish languaje. Is there anyway to download all those .gsm files speaked in spanish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Yes, there are sound sets in Spanish. Have a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international -Shaun Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] Mobile: 549 3571 543658 \ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Prompt translation: can't find please wait try ext prompt filename
On 9/19/05, Alexandre Leclerc [EMAIL PROTECTED] wrote: Hi all, I'm currently doing a french canadian (quebec) translation of the prompts. Almost all the about 140 default prompts are done, but there is one I can't find... In the directory, when the user found the good persor it press '1'. Then Please whait while I try extension... prompt is played. I can't find it and in asterisk -r mode the prompt is not displayed, but only the numbers. I tried pls-hold-while-try.gsm but this isn't it. Anyone has a clue? Is it hardcoded in the PBX? (I have version 1.0.9) It's called 'transfer' -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AstriCon 2006 Location
On 9/20/05, Darren Younger [EMAIL PROTECTED] wrote: Great Idea! I suggest Sydney :-) No complaints from me there, I live in Sydney :-) If it was in the US, I'd personally prefer the west coast. It's the easiest and cheapest part of the US for me to get to. -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Looking for firmware for Cisco 12sp+ and 30VIP
On 9/20/05, Stern, Craig [EMAIL PROTECTED] wrote: I have been looking for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any help in locating would be much appreciated. Thanks What type of firmware? The only firmware available is for SCCP/Callmanager and is a few years old. -Shaun ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] sometimes CIDNUM shows, sometimes CIDNAME??? Why, why, why, why?
On 9/19/05, Goran Dj. [EMAIL PROTECTED] wrote: Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is! Have you considered the possibility that your SIP provider may not be sending you the caller id name? CNAM looksup do cost money, and it's probably the exception rather than the norm to find a VoIP provider that will deliver it. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Callmanager Asterisk for Voicemail revisited
Some of you may remember back in May the thread on using Asterisk as a
voicemail server for a Cisco Callmanager system.
My own Callmanager system is integrated into an Asterisk server for
voicemail (and other things). Back in May I was using H323 for
integration, but since I've upgraded to CCM 4.1 I have switched over
to SIP.
The integration with H323 required using Call forwarding to send the
call to an extension on Asterisk. For example, extension 7443 would
forward to 27443 on Asterisk which looked something like:
exten = _27XXX,1,Voicemail(u${EXTEN:1})
Obviously setting this for each and every phone on Callmanager was not
an option for any wide deployments, and Paul Davidson investigated
some of the other options. Paul discovered that it was possible to
setup a voicemail pilot, tick the voicemail box, etc. but you would
lose the ability to have the caller ID information added to the
voicemail.
This wasn't an option for us, as caller ID is quite important.
Up until now, I have continued with the custom extension option,
setting up the appropriate call forwarding when new phones were added
to the system. The trunk between CCM and Asterisk changed to SIP after
the CCM upgrade, but everything else stayed the same until I revisited
this today.
To summarise what I have accomplished:
Full voicemail integration between CCM and Asterisk with the following features:
- MWI
- Voicemail on the CCM side is enabled by selecting Forward to
'Voicemail' rather than yucky custom extensions. Allows for wide
deployment.
- Messages are accessed by pressing the 'Messages' button on the CCM
phones, or dialing the VM pilot number.
- If a CCM user doesn't want to take a call, they can press the
iDivert softkey to send to voicemail immediately.
- CCM users can forward all calls to voicemail in the ccmuser pages,
or by pressing CFwdAll and entering the pilot number or messages.
- All the standard Asterisk voicemail features work just fine, eg: vm to email.
- more
Bugs with the setup:
- If there's a SIP device registered with the Asterisk machine
handling the voicemail, and the call path is something like: Sip
Device - Asterisk - CCM. If the call subsequently reaches voicemail,
Asterisk prompts for proxy authentication and CCM drops the call. This
problem can be avoided by using usernames that don't match the caller
id, eg: [sip7345], or having a machine dedicated to Voicemail.
- That's all I've found
These options have been tested with Asterisk 1.0.8 and CCM 4.1(2)sr1.
If this is something that people would be interested (and you made it
this far), I'd be quite happy to whip up some instructions and add it
to the wiki.
-Shaun
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Re: [Asterisk-Users] Semi-Ot - Cisco IP Phone Password Reset Procedure
On 7/22/05, Cory Andrews [EMAIL PROTECTED] wrote: Has anyone come up with a way to reset the password on a Cisco IP Phone when the normal password reset procedure does not work? I have some phones that are running MGCP, and the password for the phone was assigned in the original config file TFTP'd to the phone. Can you TFTP a new configuration to the phone? If it's using DHCP and doesn't have a TFTP server explicitly defined (and you have a DHCP server that supports it), you can send the phone a TFTP server address using DHCP options 66 and 150. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Asterisk and Cisco CallManager Integration
I have Callmanager 3.3(5) linked with Asterisk using H323. Note that even though I'm using Stable Asterisk from 12/06/05, I am using H323 from CVS Head 30/09/04. I found that later versions would have one-way audio problems. I've also had it working with OH323 but noticed higher audio latency (when talking between two sites over our VPN (24ms apart) people would often talk over each other). After switching to H323 (which uses Asterisk's RTP stack) this problem went away. Everything works fine, but I can't get the particular version of H323 to pass caller number and name to Callmanager (it passes one or the other). Both work from Callmanager to Asterisk though. This isn't a big deal though as most incoming calls are through the PSTN and there's no name presented. There's no gatekeeper in use - Asterisk dials directly, eg: Dial(H323/[EMAIL PROTECTED]). Callmanager does the same. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] combining calls from 2 queues
On 6/23/05, Seamus Abshere [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: this is perhaps a silly question, but how do you have so many zaptel FXS's? do you have six TDM400 cards with four FXS's each? or what am I missing? Most likely a channel bank with 24FXS. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Call Manager Asterisk for Voicemail
On 5/26/05, Scott Herrick [EMAIL PROTECTED] wrote: BUMP It's CM 3.3.6 MAN that would be sweet if * could take the place of Unity! Anybody? :-) I've got it working with Callmanager 3.3(5) and Asterisk (connected with chan_oh323). Not totally integrated - one still needs to set call forwarding (busy/no answer) on each extension that needs voicemail, but MWI works and so does the messages button (eg: on 7960G) to retrieve VM. If somebody can tell me how to send a call in Callmanager to (for example) extension 27000 when 7000 is unavailable by checking the voicemail box (rather than entering an individual number for each extension), it'll be perfect. I can share my progress so far if it will be beneficial. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Call Manager Asterisk for Voicemail
On 5/26/05, Matt Riddell [EMAIL PROTECTED] wrote: Shaun Ewing wrote: I can share my progress so far if it will be beneficial. :) Yes it would be! Okay, I've whipped up a little guide. It assumes you have a working oh323 configuration on Asterisk. I'd appreciate any feedback. It's not all that well laid out (I created it in around 10 minutes) and some if it (dialplan, etc) is specific to my configuration (four digit extensions, phones starting with 7, system features in 88xx and 89xx), but it should be enough to go by: http://asterisk.edropbox.net/ccmasteriskvm.pdf -Shaun -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Default caller ID
On 5/26/05, Tony Hoyle [EMAIL PROTECTED] wrote: Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation states that this overides outgoing caller ID and I don't want to set that (OTOH the code implies it's only a default - chan_sip.c line 4163, plus the variable name is default_callerid). The CDR records a blank callerid from the zap channel, so it's not being set to 'asterisk' there - it does appear to be in the sip code that it's being munged. I'd rather it stayed blank let the phone handle it or say something meaningful 'Unavailable','Withheld', etc. Tony In chan_sip.c, set the following: static char default_callerid[AST_MAX_EXTENSION] = Unknown; and recompile. That will make the phone display Unknown instead of asterisk. I've been doing that for around a year - works fine. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Call Manager Asterisk for Voicemail
On 5/26/05, Paul Davidson [EMAIL PROTECTED] wrote: You've done the hard bits. The bad news is that, under CCM, there's really not much in the way of VM configuration. You should set up the VM Pilot stuff to your extension for the Asterisk voicemail- this allows you to click the 'voicemail' box on each extension rather than keying it in- but you still have to touch each extension. You can use their automated tools to make systemwide changes to all extensions- but I don't trust them at all, and I don't think that would help you in this case. Yep, I've setup a VM Pilot. I changed the default pilot, so the messages key works on all phones. Phones without a mailbox, Asterisk prompts for mailbox and password. Phones with a mailbox, just the password. I'd love to see how you configured the MWI and how you've set your dialplan- from the way it looks, you're using a different extension for each mailbox. Theoretically, there should be fields on the PDUs from h.323 that show the forwarding number- that's the way Unity does it- and you go into VM for the forwarding number, not for the extension dialed. I'm not sure without playing if any of the h323 channel drivers make the forwarding number available as a channel variable- if they don't, it should be a relatively trivial patch, assuming CCM sends it across (which I'm pretty sure it does- again, time to set some debugs and watch the PDUs). The notes basically show how MWI is configured. I am actually using a different extension for each mailbox. This is something I setup a while ago to allow calls to be transferred direct to somebody's mailbox, and it has proven useful for this as well. -pbd -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voicemailbox detection:
On Apr 5, 2005 12:26 PM, Tim Connolly [EMAIL PROTECTED] wrote:
Is there any way to detect if a user has a mailbox? I want to
send all call which match _14XXX to voicemail except if the user doesn't
have a voicemail box
Have you looked at the MailboxExists app?
What you want could be done quite easily. The following is from
memory, but it should be close enough:
exten = _14XXX,1,MailboxExists([EMAIL PROTECTED])
exten = _14XXX,2,Here goes the stuff if there's no mailbox
exten = _14XXX,102,Voicemail(u${EXTEN})
-Shaun
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Re: [Asterisk-Users] Converting 7905G to SIP
On Fri, 25 Mar 2005 17:33:17 +1000, Greg [EMAIL PROTECTED] wrote: I am trying to convert my 7905G to be SIP based and seem to be running into a few hassles. Below are all the config files and logs from the server. I have tried to follow the pdf's from cisco and some posts from other mailing lists that google turnedup, but it seems that nothing is working. Am I somehow missing a fundamental step in trying to upgrade from Call Manager to SIP? Any help is greatly appreciated. Regards, Greg That's a configuration file from the 7940/7960 series phones. The 7905G uses a totally different format, has its own firmware, etc. Do you have the SIP firmware for the 7905G? -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Advanced Cisco 7960 Config
On Thu, 24 Mar 2005 21:26:27 -0800, Max Clark [EMAIL PROTECTED] wrote: Hi all, Good evening I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? The 7960 doesn't have a hot keypad (the cheaper and less featured in other ways 7905G/7912G phones do though - go figure). You need to press Speaker first. - Give the user dialtone after they dial '9'? In your dialplan, add a , after 9. eg: TEMPLATE MATCH=9,.* Timeout=3 User=Phone/ A while ago I found a cool asterisk/penguin logo to use on the phone, can anyone point me to a place I can download this again? Wouldn't have a clue, but would also like to know :) -Shaun Thanks in advance, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco 7960 SIP boot takes 2 minutes?
We have the same problem - started when we upgraded to 7.1. It isn't too much of a bother for us though, because the phones (once configured) are left alone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] 79xx 7-4
On Thu, 17 Mar 2005 08:18:30 -0500, Joseph [EMAIL PROTECTED] wrote: Mine too :) Thanks. Ditto :) -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie can't dial out to pstn
On Fri, 18 Mar 2005 12:00:58 +1000, Greg
[EMAIL PROTECTED] wrote:
Can anyone see any glaring mistakes?
Yes.
My extensions.conf part is this:
exten = _04,1,GoTo(mobile,61${EXTEN:1},1)
In Australia we don't prefix calls to mobiles with 61.
You want something like:
exten = _04,1,Goto(mobile,${EXTEN},1)
If you're using a VoIP provider that requires 61, as well as routing
calls through Zap where no 61 is required, you'll have to put in some
logic to fix that up.
-Shaun
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Re: [Asterisk-Users] ATA's
On Wed, 16 Feb 2005 07:08:08 +0800, Leo Ann Boon [EMAIL PROTECTED] wrote: See my comments in line From my experience, the ATA is a very solid, dependable piece of hardware. I was told by a source in the company that OEMs for Cisco, the units are expensive because of the high quality parts being used. The web config looks crappy but otherwise where else do you find a $100 device that does SCCP/MGCP/SIP/H323? None of the competitors even come close to that level of protocol support. For developers who have to work on various protocols, the ATA is really cool. I agree. I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. I've also tried the Sipura SPA-2000, but had some problems with it, so the Cisco ATA is my ATA of choice now. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which IP phone to use in Australia
On Wed, 16 Feb 2005 22:04:27 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: BTW, Polycom *don't* say you can't use their phone in/with any particular manner/software. All they say is that if the phone breaks, and it is caused by asterisk, then they won't really help you out. However, the fact that it works, and works well pretty much says it all. In my experience (I've had very limited experience with the cisco phones) the polycom phone is better than the cisco. They are equal in most ways, except the cisco phones require you to pay some silly licensing fee, and if you buy the phone second hand, then you can't use any firmware version without purchasing it extra At least polycom provide the software to download. Interesting. In that case, I'll bite. Anybody know of a place in Australia that sells them (preferably online)? I might look into getting a small quantity (1 or 2) to get acquanted with them. -Shaun Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Capi channel - can I route call to another channel or back to PBX and free current channel ?
On Tue, 15 Feb 2005 10:45:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi channels to route calls ? capiECT is probably what you are after. Have a look at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme Thanks in advance, regards, Rob. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Put call on hold
On Tue, 15 Feb 2005 20:44:13 +0100, Stefano Arata [EMAIL PROTECTED] wrote: Hi, I have two analog phones connected on the digium tdm22b; I can't put calls on hold by pressing the R button on the phone. I can do it only by hook flash. How can I configure asterisk to use the R button? Thank you in advance. A hook flash and recall (R) are essentially the same thing. Make sure that the phone isn't set to earth recall (it should be set to timed break recall). The tdm card might be expecting be expecting a different timed loop break value. For example, Italian phone systems expect a value of 90ms. The USA expects around 700ms. Your phone is probably configured for the Italian phone system. See if you can reconfigure the phone. Otherwise, I'm not sure if the value on the TDM400P cards can be reconfigured as I don't use them. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which IP phone to use in Australia
On Tue, 15 Feb 2005 17:13:39 +1100, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need standard handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by receptionist. The main problem is that I am in Australia and I need to get phones that can be sourced in Australia. (correct power supplies, certified for australia, etc..) I did look at supported h/w list and I am going to go through all of those companies, but I have no idea on how good/bad those phones are. I really need some advise here. The Cisco 79xx range of IP phones (including 7905G, 7940G and 7960G) work just fine in Australia, and have an A-Tick. The CP-PWR-CUBE= is the official power pack, which works with any standard IEC computer power cable. There's also a cheaper generic power pack available from some retailers, or if you have a Cisco PWR switch you can use Cisco PoE. The three types allow one to match the phone to a person's calling requirements. I typically buy my phones from Techtopia (http://www.techtopia.com.au). Buying locally means that there aren't any issues with power supplies, customs, etc. I've tried two types of phones - Grandstream BT-101 and the Ciscos. The Grandstream is useless for any serious calling, and would not be recommended for a receptionist. We've had it do all sorts of nasty things including putting a call on hold indefinitely when trying to transfer. One particular version of the firmware also caused problems with our network. It's also necessary to press send after dialing a number (or wait for the timeout). No A-Tick either. The Cisco phones do their job brilliantly - they also look very nice aesthetically. They're exceptionally easy to use with softkeys guiding one through tasks such as transfer, etc. The 7960 phone is perfect for a receptionist - it supports a headset (as does the 7940), and can have up to 12 simultaneous calls over 6 'lines'. One can also configure a dial plan on the Cisco phones, so that calls progress automatically when the correct number of digits has been entered (eg: 0 02 5551 5551). I spent around an hour testing the Polycom IP500 and IP600 phones nearly two years ago (when my interest in VoIP was picking up). They seem quite nice, but i don't have any of these deployed. I refuse to - Polycom won't sell them to people for use with Asterisk, and I'm not going to buy products from a company that try and dictate how I use their product. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which IP phone to use in Australia
On Wed, 16 Feb 2005 12:20:00 +1100, Paul Hales [EMAIL PROTECTED] wrote: Regarding your quote about Polycom - I'm not sure what you mean by 'Polycom won't sell...' We have over 100 polycom's out and about, all hooked into our 3 Asterisk servers. I will admit that I haven't enquired with Polycom, but I've read numerous times on this list and other places (can't think of references off the top of my head) that they'll only officially sell the phones if it's to be used with an approved softswitch. Not sure if that's still the case though. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which IP phone to use in Australia
On Wed, 16 Feb 2005 09:23:21 +0800, Stuart Elvish [EMAIL PROTECTED] wrote: Definitely agree - don't even try using the Grandstream for a receptionist (among other things the phone probably won't hold out physically for more than a few weeks if it makes it that far). :-) They have recently been ticked as well, plus the firmware has become some what stable that having been said I am not sure when the last update came out and it does have a couple of quirks. We have the system time out (or send the dialed digit string) after 4 seconds of no dialing which works well - but that depends on the user environment and what they expect from the phone system. The other problem is that Grandstream don't display any type of alpha caller id - they are purely a digit based caller ID presentation (it tries to present an alpha sequence but it doesn't work at all). The lack of alpha caller ID is a downside. We're using the alpha string for all sorts of things, eg: to display the trunk a call came in on Private Line, a queue QUEUE: Sales, in addition to the name of the caller where supported. It's certainly noticeable when absent. Don't get me wrong - they are still the bottom of the range / basic phone IMHO and Cisco do seem to work a lot better, but are also more expensive and my boss won't pay for one. They are more expensive, which is a downside to the Cisco phones. I bit the bullet and bought a few varying models, but it was a bit of a financial hit. I have the final say on company purchases, so there is no boss to contend with. What sort of setup is involved for the Cisco as far as config files etc? I am used to plug and play phones (Zyxel, Grandstream, HOP etc) which require minimal configuration and have no licensing issues with them. I know for the Polycom you need to get a TFTP server for XML config files running, and I believe you need something similar for Cisco phones. You'll need a TFTP server to get the SIP firmware on the phone. For small deployments you can configure the options on the phone itself, but for anything more than 2 phones, I'd recommend a TFTP server. Stuart -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Which IP phone to use in Australia
On Wed, 16 Feb 2005 13:18:20 +1100, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: You ahve to run Linux anyway. TFTP is very easy to setup. Rudolf Yep. My TFTP server is also my Asterisk server - no need for a separate machine. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] 7912G: Takes the same firmware as 7940/60?
On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet - www.acropolistelecom.net [EMAIL PROTECTED] wrote: Here it is : http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 software is the same for 7905 / 7912 It's not actually. Firmware for both versions is available from that page, but each phone has its own firmware. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Setting a Forward to an external number on your phone
On Fri, 11 Feb 2005 18:48:14 +0100, Jui [EMAIL PROTECTED] wrote: Hi! snip I am leaving my office and I want to tell asterisk to forward calls now to my mobile phone by just hitting a key (on my IP-Phone) or by using a special key-sequence. What type phones are you using? I know that with the Cisco 7940 and 7960 phones that we use, one can just hit the CFwdAll' softkey, enter a number (eg: mobile, or another extension) and press accept. To cancel call forwarding, just press CFwdAll again. Pretty simple. Caveat: This method will occupy two lines whenever a call is forwarded. If you use an analogue interface, it's also possible that the lines will be tied up forever because if they don't know the caller has hung up. The latter isn't an issue with ISDN though (which we use). -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco 7960 Beating a Dead Horse
On Wed, 09 Feb 2005 12:20:01 -0800, Max Clark [EMAIL PROTECTED] wrote: Which lead me to believe that there was an 8.3 naming issue when using a windows based tftpd server. So I changed the file names of my image to an 8.3 structure, updated the configuration files and rebooted. After I do that I see the request and transfer of the OS79XX.TXT and the SEP0007EB0630A6.cnf but nothing for the firmware, or anything about the SIP configuration files. What gives? How do I get this phone to download the SIP firmware? You need to do an incremental upgrade - eg: SIP 2.3 - SIP 4.4 - SIP 7.3. The ealier images can be obtained from Cisco if you have a valid CCO login. And yes, you're right, the default image load (SCCP) can't handle long file names which the later firmware versions use. Best, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] X100P Setup
On Wed, 2 Feb 2005 11:21:02 - (GMT), Jeff Fern [EMAIL PROTECTED] wrote: Hello all, I have just installed a Wildcard X100P into an Asterisk box. I connected the line socket to the internal telephone system where I work. The card is identified to asterisk etc, however I am unable to recieve or make calls. The PBX port it's connected to - is it on an SLT port (where any standard phone can be plugged in), or a proprietary digital port (typically where phones specific to the system plug into)? If it's the latter - the X100P won't work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Disabling native bridging for IAX calls
On Wed, 02 Feb 2005 14:02:51 +, Gareth Blades [EMAIL PROTECTED] wrote: I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does not work. notransfer=yes -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi and G711u
Hello all, I've got an AVM Fritz!Card PCI that I'm using with Asterisk under chan_capi (0.3.5) Phones on our internal network all use g711u. I'm aware the chan_capi uses g711a by default. To reduce the need for transcoding, I decided to make everything use the same codec. First I changed all the phones (Cisco models 7905G, 7940G and ATA 186 all running SIP) to g711a, but it seemed to break the echo suppressor in the ATA 186, so that wasn't a viable option. I then noticed the following in the chan_capi Makefile: # uncomment the next line if you are in the ulaw world #CFLAGS+=-DCAPI_ULAW I tried that and recompiled, but calls are filled with static and quite distorted, so it's obviously not an option. I was wondering if anybody has any tips for getting chan_capi working with ulaw, or is this feature coming later on? Thanks, Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zap channels in AU hanging up on STD pips
On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. I had the same problem (before I switched to Telstra ISDN). Increasing busycount to 8 fixed it. -Shaun Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco 7940/7960
On Tue, 25 Jan 2005 09:08:51 -0500, Mark Johnson [EMAIL PROTECTED] wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or lift the handset before you can start dialing. Is there a setting I am missing, or is this just a product of SIP and I have to live with? Thanks! Craig Unfortunately this hot keypad functionality is not included with the 7940/7960 SIP image. It is on the 7905/7912 SIP image though. As I have both types of phones in use here, it's somewhat annoying having to adjust my dialing habits depending on the phone I'm using :-) *shrugs* -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco 7940/7960
On Tue, 25 Jan 2005 17:25:30 +0200, Doug Reid - Stormcorp [EMAIL PROTECTED] wrote: We use the 7690 and it works fine there. Has nothing to do with SIP as Snom, ACT, 7960 ect all work that way. I think people are getting confused. I take it that Mark is referring to a hot keypad functionality. If you want to make a call, you don't pickup the handset, you don't press newcall, you don't press speaker - you simply start dialing the number you want. This feature is standard on some PBXs. The 7940/7960 SIP firmware does *NOT* have this functionality. The 7940/7960 Skinny firmware does. Some other Cisco phones with SIP do have this functionality, such as the 7905. As for other types of phones; the Grandstream phones don't, but apart from 2 Grandstream phones, everything here is Cisco so I'm not sure about other types. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] OT: Headset for the Cisco 7960
On Thu, 20 Jan 2005 22:19:24 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: Hello. Short of buying a (no doubt) expensive one designed specifically for the Cisco 7960, what are my options for using headset with this phone? Is there some kind of adapter to buy so I can use standard Plantronics/Jabra headsets? Is there by any chance a Bluetooth adapter - or should I just buy one of the adapters for the standard headset connector and then buy the Bluetooth adapter with those connectors? Any help would be appreciated. I use a Plantronics H51 with the Vista cord. It plugs straight into the headset port on the back of the phone with no amplifier required. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP IOS for cisco 7902G IP Phone
On Mon, 17 Jan 2005 07:11:20 -0800 (PST), R A [EMAIL PROTECTED] wrote: Hi all I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page. I need to now the name os de file or a specific category link where i can download it. If you can send me the file is beter ;-) Thanks in advance Regards Wert You're out of luck. The 7902G only supports SCCP. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP IOS for cisco 7902G IP Phone
On Mon, 17 Jan 2005 10:23:57 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page. What is IOS? Am I the only one who uses Cisco phones and doesn't know that acronym? Internetworking Operating System. It's what Cisco calls the operating system that runs on their routers, etc. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Changing caller ID based on the extension dialled?
On Thu, 6 Jan 2005 13:22:43 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: Hi list! I am going to install an intercom module for our home. The intercom can cater for 3 doors where people can ring the doorbell and the unit can also remotely open the door (Siedle system). It would be nice however to know that you are opening the right door or at which door the people are ringing. For each door I am able to program a different extension that will be dialled. Can I have asterisk change the Caller ID that will be displayed on the phone based on the extension that is dialled? The call will always be from one telephone line(extension) so this is no option. I want it like this: Door: Extension dialled Phone display (CID) Front 301 Front door (or 301) Garden 302 Garden door (or 302) Gate303 Gate I guess that this is the opposite of what Caller ID was meant for but it would suit the job that well :) Thanks!! Something like this should do the trick: exten = 301,1,SetCallerID(Front door 301) exten = 301,2,Dial phones here exten = 302,1,SetCallerID(Garden door 302) exten = 302,2,Dial phones here etc. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Special Problem in Australia ??
On Fri, 24 Dec 2004 12:17:50 +1000, Gary [EMAIL PROTECTED] wrote: Hi folks, this is specifically directed to Australia Asterisk users.. We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's). Gary I use a Telstra BRI now, but when I was using the X100P cards I had the problem. It was fixed by increasing the busycount value in zapata.conf. Anyway, I see in a later message that you've fixed the problem now. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Special Problem in Australia ??
On Sun, 26 Dec 2004 17:04:01 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Just out of interest, what BRI card are you with Asterisk? AVM Fritz!Card PCI. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Caller ID - TE405P - Telstra Onramp 10 - Australia
On Wed, 22 Dec 2004 11:53:42 +1100, Adam Goryachev
[EMAIL PROTECTED] wrote:
All looks good up until here...
Note, your phone will end up dialling which depending on your
equipment may well be interpreted as 000 and be routed to emergency...
Probably something you don't want!
Indeed.. Happened to me once.
Unfortunately I'm so used to dialing zero, I've done it from other
phones by accident. I went to make an International call, added the
zero and ended up with 00011.
IMHO, try setting it to 342 or whatever, which perhaps is an internal
extension that does a playback There was no callerid information
available for this call hangup.
I done two things:
First change (in the Asterisk source) channels/chan_sip.c:
#define CALLERID_UNKNOWNUnknown
static char default_callerid[AST_MAX_EXTENSION] = Unknown;
Then, my extension logic has:
exten = ,1,NoOp
exten = ,2,SetCallerID(Outside Call 0${CALLERIDNUM})
exten = ,3,GotoIf,$[${CALLERIDNUM} = 000]?200:4
exten = ,4,Goto(local-extensions,,1)
exten = ,200,SetCallerID(Outside Call )
exten = ,201,Goto(4)
It does two things - prefixes the caller number with '0', replaces the
name with Outside Call (so I know the call came from outside).
If it's a public number, the phones (at least those that support cid
name+number) will display Outside Call and the number.
If it's a private number, the phones display Outside Call and Unknown.
This all works fine on a Telstra Onramp/ISDN 2 (BRI). I'm not sure
about Onramp/ISDN 10/20/30 (PRI)
Regards,
Adam
-Shaun
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Re: [Asterisk-Users] Asterisk and Cisco 7905G or Cisco 7912G
On Mon, 13 Dec 2004 11:14:00 -0600, Adi Linden [EMAIL PROTECTED] wrote: Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. Both phones support SIP. I can't speak for the 7912G, but I have several 7905G phones and these work perfectly with Asterisk. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. The firmware is easy to obtain if you have a Cisco support agreement - it's downloadable from CCO (the 7905G and 7912G have different firmware builds, but a similar configuration process - be aware of that). -Shaun ___ 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] Is anyone using Cisco 7905G phones?
On Mon, 6 Dec 2004 22:45:16 -0800, Randy MacKay [EMAIL PROTECTED] wrote: I have a few Cisco 7905G phones and I having a little trouble configuring them. They are working with Asterisk. I'm able to get the sip image loaded, but I can't get the phones to blind transfer. Does the Cisco 7905G Phone use XML Services? If you are using the 7905G phone, would you post any of your configuration files so I can try and figure out where I'm going wrong? Thanks for your help, Randy I use them, they work fine with blind transfer. They don't, however, use XML services. I'll send an email following this one direct to you with my configuration. -Shaun ___ 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] Asterisk and Cisco IP Phones
On Sun, 5 Dec 2004 12:10:28 +0200, Walid Azab [EMAIL PROTECTED] wrote: Guys, obviously there is an argument about SIP vs SCCP when it comes to using Cisco IP Phones with Asterisk. I am not really sure which way to go. Probably I will go with SIP now unless you guys do recommend not using it. Walid SIP works perfectly with the Cisco IP phones and Asterisk - I'd certainly go for SIP. -Shaun ___ 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] List's quiet or down?
On Mon, 06 Dec 2004 12:34:29 +1100, David Uzzell [EMAIL PROTECTED] wrote: Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David Not just you, I didn't receive anything from the list for at least 24 hours. I seem to be receiving things now though. -Shaun ___ 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] Headsets for Cisco 7940/7960
On Sun, 21 Nov 2004 18:01:07 -0500, Brian Pavane [EMAIL PROTECTED] wrote: What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to work, however these are both over-the-head type models. I was looking for an over-the-ear model, as I would like to be able to provide a variety of headsets depending on the individuals taste. I am not looking for a headset that requires an external amplifier, but rather a headset that can make use of the headset jack on the phone itself. We use the Plantronics H51 headsets with no problems. Unfortunately (for you), it's an over-the-head type model. -Shaun ___ 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] 7912G Ringers?
On Tue, 26 Oct 2004 16:44:01 -0600, Michael Loftis [EMAIL PROTECTED] wrote: Anyone have a way to get/know if these phone support anything other than the default Chirp 1 ringer for these phones (like the 7940/7960 where you can load a fairly arbitrary number of ringers...) TIA Unfortunately they don't (same applies with 7905G). However, you can use the ALERT_INFO variable to achieve a distinctive ring. -Shaun ___ 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] Connecting to Commander NT132
On Fri, 22 Oct 2004 13:52:13 +0800, Nick Cobley [EMAIL PROTECTED] wrote: Hi, I am looking at connecting Asterisk upto a Commander NT132. I need 2 lines, and initially was going to connect it up to some analog ports, which I have since discovered they don't have. snip BTW I am in Australia if that makes a difference. It certainly does. You will only find this phone system called the Commander NT132 in Australia. Elsewhere it's known as the Nortel Norstar MICS (Commander NT40 is the Norstar CICS). http://www.voip-info.org/files/nortel-asterisk-0.2.pdf might be of some help to you. Kind regards Nick -Shaun ___ 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] SIP phones
On Wed, 20 Oct 2004 09:39:31 -0400, Michael Di Martino [EMAIL PROTECTED] wrote: I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room. Cisco 7905G, 7940G and 7960G phones have very loud ringers. -Shaun ___ 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] Intel Modem vs Digium Cards
On Mon, 11 Oct 2004 00:10:26 +0100, David J Carter [EMAIL PROTECTED] wrote: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave £55 is more like US$100 :-) ___ 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] 'asterisk' displayed on my Cisco 7960 7912...
On Wed, 22 Sep 2004 14:06:51 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert You need to set a valid caller ID number. For example, in sip.conf under the configuration for your phone: callerid=Shaun Ewing 7011 -Shaun ___ 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] SIP Phone
On Wed, 22 Sep 2004 16:40:04 +0200, Michael Bielicki [EMAIL PROTECTED] wrote: Cisco 7940 :) I'll concur with that. The Cisco 7940 and 7960 phones have great speakerphones :) As for ones to stay away from - the Grandstream BT-100 series. The sound is fine on the local end, but is very low for the remote end (sounds as if the microphone in speaker mode is actually the mic on the handset). -Shaun ___ 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] SIP Phone
On Wed, 22 Sep 2004 07:56:48 -0700, Wiley E. Siler [EMAIL PROTECTED] wrote: Do you have a price range? I don't know about pricing in the US, so I'll skip this (I buy mine in Australia). I use Polycom IP500s and the speaker phone is awesome. It picks up speakers in the room very well at 5-6 feet. Polycom has always made an exceptional speaker phone even on plain ole phones. Their implementation on the IP phones is excellent so they are my preference. The speakerphone in the 7940/7960 phones is actually made by Polycom. This probably explains why it is such good quality. I have heard that the Cisco phones are quite nice too. I think from a previous conver that the 7905 has a speaker phone and is priced fairly low. Monitor only (7905G anyway). Cheers, Wiley -Shaun ___ 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] Cisco IP phone
The 7910 does not support SIP. It is SCCP only. -Shaun - Original Message - From: Henry Devito [EMAIL PROTECTED] Date: Wed, 22 Sep 2004 10:44:02 -0500 Subject: [Asterisk-Users] Cisco IP phone To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Hi all, I have a person trying to sell me Cisco 7910 IP Phones. Does anyone know if SIP is supported on these phones? I have CCO login also so if they do support SIP does anyone know where I could download the software? Thanks in advance. ___ 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] Transfering incoming calls using same line
On Wed, 22 Sep 2004 10:20:58 -0400 (EDT), Jon Miron [EMAIL PROTECTED] wrote:
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is dialed
using the same X100P card. Basically I want to dial
in, enter 9 + phone# and have it do a flash then
have it dial *08 the same phone number + # on the
same PSTN line to have it transfer my call to another
phone number. I realize this isn't very safe, but I
would like to be able to make long distance calls to
any number while I'm out with my cell phone so I want
to take advantage of my free LD package on my PSTN
line. Thanks in advance!
Three applications that would allow you to achieve something like that
- disa, flash and senddtmf.
This is untested, but some logic like the following might help:
Use something like the following in your IVR.
exten = 10,1,DISA,1234|calltransfer
Then, add the context and code like:
[calltransfer]
exten = _X.,1,Flash
exten = _X.,2,Wait,1
exten = _X.,3,SendDTMF(*08${EXTEN}#)
exten = _X.,4,Hangup
I'm not sure off hand how you could terminate a number with #, but
DISA will proceed after 10 seconds anyway.
I'd be curious to see if it works - so let us know.
-Shaun
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Re: [Asterisk-Users] Cisco 7940/7960 and voicemailmain not able to press keys after a hold.
On Tue, 21 Sep 2004 14:35:02 -0600, James Sizemore [EMAIL PROTECTED] wrote: I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of Asterisk. I just tried changing the dtmfmode to rfc2833 for my Cisco 7940G phone and was able to replicate the problem. I've also found the same problem with my 7905G phones. The problem is fixed by changing the dtmfmode to inband on Asterisk. On the 7940/7960, you might also want to place the following in SIPDefault.cnf: dtmf_inband: 0 dtmf_outofband: avt dtmf_db_level: 3 dtmf_avt_payload: 100 ; Default 100 -- Following is for 7905G phones, but included for reference. On the 7905G it's slightly more complicated if you're not familiar with bitmap values. Change bits 4 and 5 in the AudioMode value to 0. I personally use: AudioMode:0x You will also need to change dtmfmode to inband on Asterisk was well. That also fixes the error message about RFC3389 support incomplete. -Shaun ___ 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: dial '0' for outside line and get a dialtone...
On Fri, 17 Sep 2004 13:53:11 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Maurizio Marini wrote: Thanks! That works like a charm! The only thing I'd like to do now is NOT having to press 'Dial' on my Cisco 7960 between the '0' and the rest of the number. Any options for that...? Regards, Evert Have you considered using dialplan.xml on the Cisco phones? That's what I do. I programmed the Australian dial plan in (you would setup the phone to suit your country's dial plan). The phones can produce a different dialtone upon dialing '0' by adding a comma, eg: TEMPLATE MATCH=0,02 Timeout=0 User=Phone/ -Shaun ___ 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] Wall-mounting UIP 200 and SoundPoint IP600 keeps coming off hook
The Cisco 7905G phones can be mounted on a wall quite easily. They also support PoE (Cisco PoE; http://www.voip-info.org/tiki-index.php?page=Cisco%20POE might be useful). -Shaun - Original Message - From: David Gomillion [EMAIL PROTECTED] Date: Fri, 3 Sep 2004 16:41:29 -0500 Subject: [Asterisk-Users] Wall-mounting UIP 200 and SoundPoint IP600 keeps coming off hook To: [EMAIL PROTECTED] I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver on-hook? Mine keeps being pushed up by the little piece of plastic that is supposed to detect if it's on-hook. It looks like the handset already has the little hole for the hook, but I didn't find said hook in the package. Has anyone else had this problem/found a solution? Thanks, David Gomillion ___ 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] Cisco 79XX SIP Ring Tones
On Tue, 31 Aug 2004 19:01:06 -0500, Christopher L. Wade [EMAIL PROTECTED] wrote: Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. It works just fine here. I have Asterisk installed in my home office. I have a residential/private number and the business numbers. I like the private number to ring to the phone on my desk, so I use the ALERT_INFO to change the ring cadence. Basically, this is what I do: exten = xx,1,SetVar(ALERT_INFO=Bellcore-dr2) exten = xx,2,Dial(SIP/7011|15|r) exten = xx,3,Voicemail,uxx Possible values are Bellcore-dr1 through Bellcore-dr5 if memory serves. -Shaun ___ 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] OT: Headset for Cisco 7960?
On Tue, 31 Aug 2004 15:58:16 -0500, B. J. Bomar [EMAIL PROTECTED] wrote: I use a Plantronics Supra H51 plugged straight into the headset port, and it works great. B. J. Same here. They're wonderful headsets. -Shaun ___ 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] Voip phones headsets
I don't have pricing, but I'm using Cisco 7940G phones with Plantronics Supra headsets and they work perfectly - no amp required either. Same story with 7960G The good thing about the Cisco phones is that you have 3 options - handset, headset or speaker. Plenty of other phones require you to pick up the handset to answer a call with your headset and you can't switch between them without swapping cables. -Shaun - Original Message - From: neil [EMAIL PROTECTED] Date: Wed, 25 Aug 2004 21:38:44 +0100 Subject: [Asterisk-Users] Voip phones headsets To: [EMAIL PROTECTED] Hi all, I wonder if anyone could recommend a voip phone that supports headset working which works with * and advise me of a supplier of same. If any suppliers wish to respond please do with pricing for 60 phones shipped to the UK. Thanks in advance Neil ___ 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] Cisco 7940 Question
On Mon, 23 Aug 2004 09:07:26 -0500, Christopher L. Wade [EMAIL PROTECTED] wrote: Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle. I'm just trying to nail down the exact number when the phone is only assigned one directory number (extension). The Cisco 7940 can handle 2 directory numbers, not one. The phone can handle 2 simultaneous calls per line/DN plus an extra call on each line for call transfer. Thanks, Chris -Shaun ___ 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] Pulse dialed digit recognization
On Sun, 22 Aug 2004 23:07:04 -0300, Daniel Bichara [EMAIL PROTECTED] wrote: Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a pulse dial telephony. Could * detect digits dialed on pulse telephones? I don't know of any IVR system that could detect pulse digits as pulse/decadic operates using electrical pulses rather than audible tones. What you should do is have a timeout. If your menu/IVR doesn't detect a dialed digit within x seconds, then send them to the operator. For example, I have: exten = t,1,Goto(callqueues,8504,1) ; direct to operator on timeout exten = s,1,DigitTimeout,5 exten = s,2,ResponseTimeout,10 exten = s,3,Background,ivr-welcome Rest of options... Daniel -Shaun ___ 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] Does Granstream BT100 Conference Button Work?
On Thu, 19 Aug 2004 15:53:38 -0400, James Freire [EMAIL PROTECTED] wrote: Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James I thought it was there for decoration? ;-) It does absolutely nothing on my BT-101 phones. -Shaun ___ 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] PoE injectors
Have a look at http://www.voip-info.org/tiki-index.php?page=Cisco%20POE -Shaun - Original Message - From: Gonzalo Gasca Meza [EMAIL PROTECTED] Date: Fri, 20 Aug 2004 21:57:01 -0700 (PDT) Subject: [Asterisk-Users] PoE injectors To: [EMAIL PROTECTED] Anyone knows some home-use PoE injector that works ok with Cisco 7960s? Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. ___ 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] Isdn4Linux and DTMF
Hello all, I currently have an Eicon Diva Client isdn card using i4l. Outbound dtmf doesn't work (and never has), but there has been an annoying problem with false dtmf detection in calls (that could be triggered easily by blowing into the receiver on the remote end). I looked through the list and found two patches that need to be applied - 1 to isdn_tty.c in the kernel, and another to chan_modem_i4l.c in Asterisk. This morning (and I say morning - because it's now 4:50am Friday here) I was feeling adventurous, so I applied the patches and recompiled my kernel. Once that was done, I started up Asterisk and noticed that there was no false DTMF (well - no dtmf was detected at all). Time for the second patch. I applied the second patch to chan_modem_i4l.c successfully and recompiled Asterisk (make clean; make; make install). Unfortunately I still can't get DTMF working. I've included my modem.conf below in case there might be any settings that could help: [interfaces] context=incoming-isdn driver=i4l language=en type=autodetect dialtype=tone mode=immediate features=noquelch,dtmf group=1 msn=4627 incomingmsn=4625,4625,4627 device = /dev/ttyI0 device = /dev/ttyI1 If anybody knows anything that might help - any help would be greatly appreciated. Failing that - does anybody know where to buy an AVM Fritz!Card in Australia? (or some card that plays nicely with ISDN and Asterisk and doesn't use i4l)? I found one place that sells them but the price is more than I was hoping to pay (when looking at what they're available for from Germany, etc). Thanks, Shaun ___ 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: Re: [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)
Thanks for your advice everybody (replying collectively). Now I'm in a bit of dilemma. I'm not reselling the system, it's all for my home office (where my Asterisk install is). I've sent off a couple of emails, so I'll see what happens :-) Thanks again, Shaun ___ 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] 7960 help
On Wed, 18 Aug 2004 08:49:49 -0400, Donald Hall [EMAIL PROTECTED] wrote: The problem appears to be that a 7960/7940 running P003AM30, the load shipped from the factory, cannot load a new load file that is more than 393216 bytes in size. Easily solved. Load an older version like P0S30203 on it, and then upgrade to the larger and more recent SIP images. -Shaun ___ 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] Cisco 7.2 firmware for SIP 7940/7960 released
Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to know. -Shaun ___ 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] Cisco 7.2 firmware for SIP 7940/7960 released
On Tue, 17 Aug 2004 15:33:40 -0400, Wojciech Tryc [EMAIL PROTECTED] wrote: What's the URL. I have 7960 with the old firmware, it works fine..but I wouldn't mind to update to the latest/ Wojtek http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 It's also on their FTP - ftp://ftp.cisco.com/cisco/voice/ip-phone/sip-7960 You will need a Cisco customer login for both of those. -Shaun ___ 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] The CISCO 7940 Tranfer Button..
On Wed, 11 Aug 2004 02:46:14 -0700, lists-jmhunter [EMAIL PROTECTED] wrote: also frastrated by this... 7905g are laid out a lot better Irrelevant - you still need to press More to get transfer on the 7905Gs as well. I have Smartnet on my phones - I wonder if it's worth putting in a feature request to Cisco? *shrugs* -Shaun ___ 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] Called ID in Australia
On Tue, 3 Aug 2004 20:24:50 +1000, Robert Barnes [EMAIL PROTECTED] wrote: Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes Australia uses the bellcore caller ID standard which is the same as that used in the USA, Canada, and a few other countries. Depending on the hardware you use, caller ID should work out of the box - at least it did with my ISDN and X100P cards. -Shaun ___ 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] New Zealand DIDs
On Thu, 29 Jul 2004 09:43:18 +1000, Greg Hulands [EMAIL PROTECTED] wrote: What about Australia? Greg http://www.atp.org.au We've been using them for a few months now. -Shaun ___ 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] hold then transfer...
What phones and Interface are you using? -Shaun - Original Message - From: Stephen Hon [EMAIL PROTECTED] Date: Tue, 20 Jul 2004 12:30:29 -1000 Subject: [Asterisk-Users] hold then transfer... To: [EMAIL PROTECTED] Hi.. Has anybody been experiencing any problems with transfers using # after holding? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the new call, places new call on hold, resumes old call and tries to transfer won't work. At first, I thought somewhere along the line the 'Tt' options must be messed up in a dial command somewhere.. but I double checked everywhere and ensured that I was enabling transfers. Does anybody have any suggestions? Thanks, 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
Re: [Asterisk-Users] PRI dead in USA?
On Tue, 20 Jul 2004 11:46:15 -0500, brian [EMAIL PROTECTED] wrote: Well they fail to realize that ISDN is used for more than data. I just wanna scream at them and say IT DOES VOICE TO YOU NINNY!.. Rates are far from reasonable. 167/mth here is what I would have to pay for ISDN-BRI. SBC is lame. bkw Fun isn't it. I recently had a BRI installed and had to firmly tell the operator that I didn't want an Internet service to go with it. Conversation goes something like: Me: I'd like to order an ISDN line please Operator: ISDN is a last resort option for those who can't get broadband Internet. Me: I want to use it for voice. Operator: Do you want an Internet plan to go with that? Me: No. I already have ADSL on another line. Operator: Why do you want ISDN then? Me: Because it's cheaper and better than having two POTS lines. Operator: Orders ISDN Conversation goes on and the Operator asks me again if I want an Internet plan. Anybody would think that they didn't want to sell ISDN services even though the telco (Telstra) has recently ran a big promotion pushing the benefits of ISDN. I'm paying AU$45.50 per month including tax for my BRI, which is around US$33. -Shaun ___ 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] CISCO 7960G FIRMWARE
Nonsense. If you have access to the firmware, they're fantastic phones and the best phones I've ever used. If you buy the phones new from an authorised dealer, buying the Smartnet contract will cost a few dollars and only take a couple of days to process giving you full access to Cisco's website (including the firmware). -Shaun On Wed, 14 Jul 2004 17:19:10 -0400, Kanuri, Seshu [EMAIL PROTECTED] wrote: Hi All, CISCO 7960G is may not be a good decision. Without Call Manager, this is garbage. A friend of mine bought this a couple of days ago and had to return quickly, to get the refund. Seshu Kanuri ___ 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] CISCO 7960G FIRMWARE
On Thu, 15 Jul 2004 10:08:41 +0200, Michael Devenijn [EMAIL PROTECTED] wrote: --- example B --- You first have to push new call before dialing a number. this seems to be a detail but 1° why ? and 2° try to migrate users from another system ... I don't know how your handsets are setup, but I have *never* had to do that with my 7940/7960 phones (unless using speakerphone, in which case it's either New Call or the speaker button). I simply pick up the phone and dial away. Because I have a dialplan setup, the phones sense when enough digits have been dialed and sends the call immediately. All people need to remember here is to dial 0 before the external number. My 80 year old Grandmother has even used the Cisco phones with no problems, and she hasn't even used a computer before. -Shaun ___ 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] Cisco phones and Messages and Forward ToVM keys
The 7905G phones with the SIP firmware have this feature.
-Shaun
On Fri, 16 Jul 2004 01:14:28 +0100, Wayne [EMAIL PROTECTED] wrote:
Hiya,.
What cisco phone / firmware are you using? - Ive got a 7960 with SIP and
the only 'soft' button that comes up when you get an incomming call is
to 'answer', havnt got an option for 'send to VM'
Thanks
Wayne.
Brian wrote:
; Below assumes you are using the same number for Voicemail boxes as
extensions
; if ${RDNIS} is blank then GotoIf will go to extension 2, otherwise it
will go to extension 102
exten = 8500,1,GoToIf($[X${RDNIS} = X]?2:102)
exten = 8500,2,VoiceMailMain(s${CALLERIDNUM})
exten = 8500,3,Hangup
exten = 8500,102,VoiceMail(u${RDNIS})
exten = 8500,103,Hangup
; you should now be able to press the Messages key and get Voicemail
man, and press the ToVM key when you have an incoming call and have
the call immediately forwarded to your voicemail with the unavailable
greeting.
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Re: [Asterisk-Users] Onhold Music
killall -9 mpg123 That works for me (eg: if I add new music, etc). The music should loop, so if it's not doing that for you, you have a problem :-) -Shaun On Wed, 14 Jul 2004 09:43:06 -0400, Joseph [EMAIL PROTECTED] wrote: Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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] CISCO 7960 VLAN
As far as I know, if your switch doesn't support CDP, you need to configure the VLAN on the phone. It's in Settings - Network Configuration - Option 22 Admin VLAN Id. You will need to unlock the configuration first (method depends on the SIP firmware version you have). -Shaun On Tue, 13 Jul 2004 01:16:03 -0400, Kevin [EMAIL PROTECTED] wrote: I noticed in the Cisco documentation that the access port( the port to hook to a PC) on the 7960 can be configured via CDP with a layer3 Cisco switch. I also see where in the SIP configuration that you can specify the ADMIN VAN. Does anyone know to configure the 7960 access port to use a different VLAN using a non Cisco switch? Thanks, Kevin ___ 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] Help Needed in configuring Cisco 7940
The configuration for the 7940 is the same as the Cisco 7960, look at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx and http://www.voip-info.org/wiki-Setup+SiP+on+9740+-+9760. Quoted from the wiki: Note: Cisco software images are only available from Cisco's web site and are protected by copyright laws. Access to their web site requires an account be established. The easiest way to do that is to purchase a Maintenance Agreement from Cisco for approximately $8 per year (US). It's the only way you will be able to get access to the images unless somebody sends them to you illegally. As somebody who has gone down the maintenance (Smartnet) route, it's certainly the best way. -Shaun On Tue, 13 Jul 2004 08:14:07 -0700 (PDT), oi geli [EMAIL PROTECTED] wrote: I bought a Cisco 7940, I need to configure it for Asterisk. I checked the wiki pages. Followed the link to Cisco web page. Tried to download the image for SIP. It wo'nt allow me even though I registered for the CCO Valet. Is the image available anywhere else? I saw some of the messages in the mailing list that it supposed to be fairly simple. I would highly appreciate if somone could post the step by step configuration process in detail. Thanks in advance.. __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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] How to make * don't strip the leading 0
On Mon, 12 Jul 2004 14:57:42 +0200, Kai Militzer [EMAIL PROTECTED] wrote:
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile phone
0177-1234567 should be displayed, but 177-1234567 is displayed. I double
checked if I've forgotten to remove an option to strip the first digit
of incoming calls and found nothing.
The wiki and the mailinglist archives can't enlight me either, why
asterisk behaves like this, or how I can turn it off. So if someone
could give me a hint, I would be very delighted!
You could try adding the leading zero.
For example, I have:
[incoming-isdn]
[incoming-isdn]
exten = msn,1,NoOp
exten = msn,2,SetCallerID(0${CALLERIDNAME} 00${CALLERIDNUM})
exten = msn,3,GotoIf,$[${CALLERIDNUM} = 000]?200:4
exten = msn,4,NoOp
exten = msn,5,Goto(local-extensions,7000,1)
exten = msn,200,SetCallerID(Private )
exten = msn,201,Goto(4)
(my number has been replaced with msn)
This adds the leading 0 to calleridname, and 00 to calleridnum (so it
included the '0' needed to dial externally). It has an unfortunate
side effect of setting the caller ID number to '000' if the telco
doesn't send any caller ID (which also happens to be the emergency
number here in Australia), so I have the GotoIf to catch that
condition and replace it with Private.
I don't know how that works for incoming International calls (never
tested), but it works just fine for national calls.
-Shaun
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Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Tue, 6 Jul 2004 00:32:20 -0500, Chris Foster [EMAIL PROTECTED] wrote: stable's gone because it wasn't too stable. The lastest CVS source is alot more full featured and stable then the old stable branch. I've found it the opposite. I've tried CVS Head a few times because I wanted some of the latest features, but every time I have gone back to the trusty Stable CVS build we've been using on two machines. Not once has CVS Stable gone down or had any problems. -Shaun ___ 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: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
On Mon, 05 Jul 2004 22:38:22 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote: Hall, Eric M. wrote: I'm trying to see if this is even possible. AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk, the call is complete and answered when it starts ringing. A PSTN/POTS call is always going to be the final destination. With Analogue interfaces (X100P, etc) - yes, a call is marked ANSWERED as soon as it starts ringing. It's a different story with ISDN/digital interfaces. On several occasions I've set my desk phone to ring with my cell phone, etc. - the first one to answer gets the call. -Shaun ___ 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] What happened to the CVS asterisk_stable branch?
On Tue, 6 Jul 2004 00:18:15 -0700 (PDT), every buddy [EMAIL PROTECTED] wrote: That's good news. Unfortunately I don't seem to have any record anymore, Always looked it up on the web site. Would you care to post here what the command was again to get the stable branch from CVS? thanks. It was: cvs checkout -r v1-0_stable asterisk So, assuming the tag is the same, that should work. -Shaun ___ 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] 2x analog interface (1 ISDN and 1 door phone) recomendation for Europe ?
http://www.voip-info.org/wiki-Asterisk+phone+door might be of some use. -Shaun On Tue, 6 Jul 2004 09:27:07 +0200, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert. ___ 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] How do I disable '#' to transfer a call?
Easy, just don't include t or T in the dial string options. -Shaun On Tue, 06 Jul 2004 01:38:23 -0700, Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote: I don't see anything on the Wiki or in the documentation about disabling this feature. ___ 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] What happened to the CVS asterisk_stable branch?
On Tue, 6 Jul 2004 22:43:41 -0400, Leif Madsen [EMAIL PROTECTED] wrote: Pssst... information was also available at http://www.asteriskdocs.org :) http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x251.html Leif Madsen http://www.asteriskdocs.org BTW: We can *always* use more help documenting... Good point. My C knowledge is very basic (ie: I could write basic programs, and make simple modifications, but that's about it), so I can't contribute that way, but documentation is something that I can do. Anyway, it's half way through the business day here in Australia, so I must leave this email there - but I'll look into it later. -Shaun ___ 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
