[asterisk-users] analog fax extension dialing out
I would like to setup my fax extension through freepbx to NOT have to dial 9. I will never dial internal numbers, so all I want it to do is pass the digits to the trunk. Is that possible with freepbx and if so, how is it accomplished? Thanks in advance Sean Garland, V.P. Siskiyou Technology Consultants 510 N. Mt. Shasta Blvd. Suite D Mount Shasta, CA 96067 ph/fax: 530-926-1489 http://www.siskiyoutech.com [EMAIL PROTECTED] ___ --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] Phones that work well through NAT
So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, April 16, 2006 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones that work well through NAT I'm really not interested to look back, but IIRC, when using just one Polycom phone behind NAT we didn't have any problems, but when using more than one behind the same NAT that is when problems started, qualify=somethingbutno seemed to help it a bit, but didn't eliminate the problem. On 4/16/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/313 - Release Date: 4/15/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
So I have * box shorewall/linux NAT firewall internet - WRT54G with openwrt - IP500 I have 5060, 4569, and 1 through 2 forwarded to * box from internet. I have tried everything I can think of on the wrt to get it to work but it appears, looking at tcpdump that my phone is trying to get to the * box (I can get one way audio with port mapping in the WRT) using the 192.168.x.x address it has as its internal interface... Is there a way to force the IP500 to use the public IP of the * box for RTP? Then it should work... Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, April 18, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Phones that work well through NAT On Tuesday 18 April 2006 09:57, Sean Garland wrote: So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. You don't do anything to get it to work through NAT. If your * box is behind NAT you need to screw around a little, but for situations like this: * box --- [internet] --- [nat dsl router] --- IP501 all you do is set 'nat=yes' on the * box, in the IP501's peer setting. That's it. It even works with multiple IP501s behind the same NAT DSL router. If you have a stupid NAT box that closes ports off too quickly or plays too many games with the packets you may need some additional configuration (shorter registration expirations, etc.) but just buy a decent NAT box... WRT54Gs work just fine in their default configuration, for example. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] freepbx dialing prefix
Just an update found a few bug tickets regarding it and a change to page.trunk.php which allows the w. Apparently it will be fixed by version 2.1 Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, April 12, 2006 9:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] freepbx dialing prefix Submit a bug report to the FreePBX team? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Wednesday, April 12, 2006 8:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] freepbx dialing prefix I need to put a w in the dialing prefix, but it says it isnt valid. If I manually modify the extension file, it then affects all calls made over any trunk. Any ideas? Sean -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/310 - Release Date: 4/12/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] freepbx dialing prefix
I need to put a w in the dialing prefix, but it says it isnt valid. If I manually modify the extension file, it then affects all calls made over any trunk. Any ideas? Sean -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beeps and noises during calls
This hit it right on the head - the tdm card was sharing irq with the nvidia and yukon lan adapters. What a pain it was to get them off - had to trial and error the position of my raid card and tdm card and disable everything. I think its fine now, could not reproduce the problem tonight. To anyone experiencing issues like this make sure you check irq sharing. I have been dealing with this for quite some time, and getting the tdm card on its own irq, it is now working correctly and quietly on an ASUS mobo. Thanks again guys!!! Sean Garland Siskiyou Technology Consultants Mount Shasta, CA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Friday, April 07, 2006 6:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Beeps and noises during calls Try going through this PCI bus troubleshooting guide. http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting -Original Message- From: Sean Garland [mailto:[EMAIL PROTECTED] Sent: Friday, April 07, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Beeps and noises during calls Sounds like it might be the pci bus.. I have a single tdm400 card and it isn't sharing an irq with other devices. So that leaves the pci bus. Weird that I would get it from 2 separate computers though and different cards (had s100u's before). The mobo is an ASUS A7N8x-E deluxe, with Nforce 2, Althlon xp 3200+ and gig of ram... Guess I could replace the box with other hardware. I think I have another box here and I still have the s100u cards, maybe I'll put together something else to see if there is a difference... Any other ideas would be great. Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, April 07, 2006 12:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beeps and noises during calls On Friday 07 April 2006 15:03, Sean Garland wrote: The beeps are not DTMF tones (at least they don't sound like it). It sounds more like the system is trying to compensate for something or adjusting something. There is a beep, sometimes several, or maybe one or 2 in a row, and it can be faint, or loud, or whatever, but is always the same pitch and tone. Sometimes it is accompanied with loud talkback to the earpiece. I'm going nuts, and cannot in good conscience, install or recommend this to anyone till I can resolve this. It has Sounds like the system is either sharing interrupts or the system has a REALLY crappy PCI bus. I ran across this on two motherboards, one of which was really suprising because it was a decent vendor (Asus) and wasn't doing anything other than Asterisk. You don't need shared interrupts to get this. I had issues with a Sangoma A101u and Sangoma S518 in the same box (cheapass Dell P3) -- they were not sharing interrupts but the T1 would have all kinds of glitches JUST like you describe. Put a Digium T100P in place of the A101u and it worked great. (Sounds counter to the typical threads here, but it's the truth, I swear.) Again, these two cards were NOT sharing interrupts with each other or any other devices on the system. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.0/306 - Release Date: 4/9/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beeps and noises during calls
I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beeps and noises during calls
The beeps are not DTMF tones (at least they don't sound like it). It sounds more like the system is trying to compensate for something or adjusting something. There is a beep, sometimes several, or maybe one or 2 in a row, and it can be faint, or loud, or whatever, but is always the same pitch and tone. Sometimes it is accompanied with loud talkback to the earpiece. I'm going nuts, and cannot in good conscience, install or recommend this to anyone till I can resolve this. It has happened with 2 separate installs of *, with different hardware, different packages installed (one is * 1.2.4 with freepbx, the other was * 1.0 with nothing), and different digium hardware. The only thing that was the same is the Polycom phones, and SBC as a provider for the POTS lines... HELP! Thanks Sean Garland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoa Sent: Friday, April 07, 2006 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Beeps and noises during calls Some more info: TALKOFF is the wrong recognition of DTMF component in human voice as true DTMF signal. This is an unavoidable factor since human voice always contain valid DTMF combination. Fortunately, presence of these valid DTMF components are unsteady. Unlike real DTMF generated from a touch-tone keyboard, these 'human' DTMF cannot maintain on a constant combination. So they can be isolated by DELAY discrimination. If a decoded DTMF signal can stay on constantly for certain duration which exceed those normal period experienced in human voice, then it can be identified as a real DTMF command. taken from: http://www.qsl.net/ve3rgw/dtmfsql.html Zoa wrote: Have a look at this : http://www.dslreports.com/forum/remark,9151528 If anybody would have such a mitel or bellcore dtmf talkoff wav file, i have a very big email box you can drop it in :p Zoa Sean Garland wrote: I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beeps and noises during calls
Sounds like it might be the pci bus.. I have a single tdm400 card and it isn't sharing an irq with other devices. So that leaves the pci bus. Weird that I would get it from 2 separate computers though and different cards (had s100u's before). The mobo is an ASUS A7N8x-E deluxe, with Nforce 2, Althlon xp 3200+ and gig of ram... Guess I could replace the box with other hardware. I think I have another box here and I still have the s100u cards, maybe I'll put together something else to see if there is a difference... Any other ideas would be great. Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, April 07, 2006 12:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beeps and noises during calls On Friday 07 April 2006 15:03, Sean Garland wrote: The beeps are not DTMF tones (at least they don't sound like it). It sounds more like the system is trying to compensate for something or adjusting something. There is a beep, sometimes several, or maybe one or 2 in a row, and it can be faint, or loud, or whatever, but is always the same pitch and tone. Sometimes it is accompanied with loud talkback to the earpiece. I'm going nuts, and cannot in good conscience, install or recommend this to anyone till I can resolve this. It has Sounds like the system is either sharing interrupts or the system has a REALLY crappy PCI bus. I ran across this on two motherboards, one of which was really suprising because it was a decent vendor (Asus) and wasn't doing anything other than Asterisk. You don't need shared interrupts to get this. I had issues with a Sangoma A101u and Sangoma S518 in the same box (cheapass Dell P3) -- they were not sharing interrupts but the T1 would have all kinds of glitches JUST like you describe. Put a Digium T100P in place of the A101u and it worked great. (Sounds counter to the typical threads here, but it's the truth, I swear.) Again, these two cards were NOT sharing interrupts with each other or any other devices on the system. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] random beeps during calls
Asterisk 1.2.4 FreePBX 2.0.1 I am running a TDM400 card with 3 FXO and 1 FXS cards. During most all calls, there are random beeps in the background. The other party cannot hear this (I believe since they haven't said anything). This will happen when people leave voicemail, and also on most or all calls. I am running Polycom IP 501 phones. The beep/tones are not DTMF. It does not happen with a particular anything. Just place a call and while on the phone, you will often hear little beeps in the background. Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem loading zaptel drivers
I was running asterisk 1.0 and amp and tried to update tonight. Now I cannot load any zaptel drivers, I get the message module wctdm not found. Im running it on Mandrake 10.1 (2.6 kernel). HELP!!! Thanks guys Sean Garland ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to buy POLYCOM phones?
I bought mine from neutronexpress.com. Paid a little more then the voipsupply.com, cuz I didn't know about them... Good to see another place to purchase the phones... Got the firware from a poster on this list... Thanks Sean Garland Siskiyou Technology Consultants -Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Monday, October 18, 2004 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Where to buy POLYCOM phones? Either way. I've bought several devices from b2tech on ebay as well as several devices direct from voipsupply.com so it wouldn't sway me much if they were plugging their own company on this list, I already trust them. Never bought Polycom from them though, although I plan to in the near future. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Monday, October 18, 2004 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? Sales Department wrote: Try www.VOIPSupply.com they have the model 300, 500 and 600 phones available. Oh they do. I think that you mean we do. In the future, when you plug your own business you should probably mention it. whois voipsupply.com: Organization: b2 Technologies Cory Andrews 454 Sonwil Drive Buffalo, NY 14225 US Phone: 716-630-1555 Fax..: 716-630-1548 Email: [EMAIL PROTECTED] Registrar Name: Register.com Registrar Whois...: whois.register.com Registrar Homepage: http://www.register.com Domain Name: VOIPSUPPLY.COM Created on..: Mon, Apr 19, 2004 Expires on..: Wed, Apr 19, 2006 Record last updated on..: Wed, Jun 02, 2004 Administrative Contact: b2 Technologies Cory Andrews 454 Sonwil Drive Buffalo, NY 14225 US Phone: 716-630-1555 Fax..: 716-630-1548 Email: [EMAIL PROTECTED] Technical Contact, Zone Contact: Register.Com Domain Registrar 575 8th Avenue - 11th Floor New York, NY 10018 US Phone: 902-749-2701 Fax..: 902-749-5429 Email: [EMAIL PROTECTED] Domain servers in listed order: NS1.B2LLC.COM 24.75.56.200 NS2.B2LLC.COM 24.75.46.242 -- Kristian Kielhofner ___ 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 ___ 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] Echo problems polycom and x100p
Title: Echo problems polycom and x100p I am having a persistent echo problem with my polycom sip IP 500 phones. I have two x100p cards for the analog phone lines to come into the * box. The echo is very slight sometimes, to very noticeable. Also, there is often some beeps in the background at random times. The beeps can also be heard on messages which leads me to believe its the x100p cards or something For those of you that have a fully functional system with polycom ip phones, what were the settings that worked the best to cancel the beeps and echo? Thanks a ton Sean Garland Siskiyou Technology Consultants ___ 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] Echo on polycom sip phone
Title: Echo on polycom sip phone I have 2 of the X100P cards in my Mandrake 9.1 box, and 3 Polycom 500 phones. I have a terrible echo problem. It will be fine and then while you are talking it gets really loud and distorted and then will die down again. The machine is a Duron 750 with 128 MB ram, is that enough? I am hoping that its my machine and upgrading will solve it. But if not, then I have a real problem. HELP! Thanks Sean Garland Siskiyou Technology Consultants s ___ 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] Emailing phone messages?
Where do you set the outgoing mail server for use with asterisks mail system? I have entered the info in the voicemail.conf file correctly, but I am still unable to get the voicemail messages via email. I ran a tcpdump on the system while calling in and leaving a voicemail and I don't even see the system try and contact a mail server. HELP!!! Thank you all in advance. Sean Garland Siskiyou Technology Consultants ___ 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] Voicemail problem
How do you specify sendmail, or any mail program? I changed the servermail= to equal my in-house exchange server, and allowed relaying by it's the pbx's IP address, but I still don't understand how it know where to send or what program it uses.. Thanks Sean -Original Message- From: public [mailto:[EMAIL PROTECTED] Sent: Friday, June 11, 2004 2:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail problem Sean, I use the sendmail app on the pbx itself (redhat 9.1) with the serveremail=localhost Not a lot of overhead on this process, of course sendmail needs to be able to route to the internet to send out mail, so this can't be a private subnet only pbx. -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Friday, June 11, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail problem I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 = 1234,Sean Garland,[EMAIL PROTECTED] 101 = 1234,Jason Madden,[EMAIL PROTECTED] 102 = 1234,Melinda Garland,[EMAIL PROTECTED] Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.701 / Virus Database: 458 - Release Date: 6/7/2004 ___ 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
[Asterisk-Users] Voicemail problem
I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 = 1234,Sean Garland,[EMAIL PROTECTED] 101 = 1234,Jason Madden,[EMAIL PROTECTED] 102 = 1234,Melinda Garland,[EMAIL PROTECTED] Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ 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] This is a test
I am testing my spam filter to see if it is still catching all of the mailing list stuff. Thanks. Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] ___ 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] Polycom phones noise cancellation
In almost all my calls now, I am getting beeps and loud and soft parts of a conversation. It is getting very irritating. Has anyone had this happen? How do I get rid of it? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telemarketer handling
Occasionally I get calls that register asterisk on the caller id, which I am assuming means that the caller id info is not there. Is there a way to have those calls route through IVR so I don't have to deal with them? Typically they are sales calls. Thanks Sean Garland, ___ 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] Direct mailbox transfer
Thanks guys I will try that in the morning...
Sean
-Original Message-
From: John Fraizer [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 12, 2004 5:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Direct mailbox transfer
Sean Garland wrote:
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able
to transfer the call directly into a persons unavailable mailbox.
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
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Add the following context and make sure it's available to the person who
will be transferring people:
[direct-vm]
exten = _*9.,1,Voicemail2(u${EXTEN:2})
exten = _*9.,2,Hangup()
To transfer someone straight to VM, they simple blind transfer them to
*9[voicemail extension]
Works like a charm.
John
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[Asterisk-Users] Direct mailbox transfer
How would one implement a direct mailbox transfer using the macros? What I want to do is have the person who answers the call to be able to transfer the call directly into a persons unavailable mailbox. Thanks Sean Garland, MCP+I, A+ Siskiyou Technology Consultants ___ 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] Sip transfers
My polycom phones have a transfer button on them and it used to work, now it puts the call on hold, you are allowed to call the other extension and tell them the call is there, but when you hang up, the call stays on hold, and the extension you are trying to transfer to gets nothing. Ideas? Thanks Sean Garland [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parking extension not working
Okay, lets start over... I have basically a simple parking.conf, I use 701 as the parkext because of Andy Powells suggestion, I use a range of 702-710. I have in my [sip] context and include = default and in the [default] context, I have the stock include = localcalls which has on include = parkedcalls. When asterisk is loaded and I type show dialplan at the console, I get a list of what appears to be all the extensions available to me and their respective contexts. Nowhere in that list is an extension 701. Now when a call comes in, my Polycom phone rings and I answer it. If I want to park the call I press the conference button, and type 701, which then gives me a busy signal for a few seconds, and the call is terminated. I ran a sip debug and did the same procedure, and I can see that the polycom phone requests extension [EMAIL PROTECTED] but then there is a 404 not found error. Am I doing something wrong in the transfer procedure? When I dial # on the polycom phone I get an instant busy, which I assume means that my phone doesn't like that digit. So maybe my question should be, is anyone using polycom sip phones and able to use the parked call feature? Sean -Original Message- From: Lance Arbuckle [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Parking extension not working Sean Garland wrote: Yes, I have the include = parkedcalls in the default context which is where my calls start from, in fact if I try to transfer to one of the parked locations as a test of transferring to the 7xx area, I get the ...no parked call.. message so it seems like the context is working. All of the parking location extensions show up in the dialplan. What should I see on the console when transferring to 700 (or 701 as it may be) from SIP? Will it work from sip? Is there a way I can test from the console? Thanks Sean I don't think you can transfer to a park extension directly. In other words, you transfer to the parking lot ( default exten 700 ) and Asterisk parks the call in whichever parking space is available. Then Asterisk reads back the number of the parking space. On my system, with 700 as the park extension and 701-720 as parking spaces, I get the message Sorry, there is no call parked on that extension if I try to transfer to 701 which is what I would expect since it's only exten 700 which has the special park functionality. -Lance ___ 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
[Asterisk-Users] Analog phone transfer
I have the Digium usb FXS device and an analog phone attached. How do I transfer calls? Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog phone help
I have 2 sip phones and an analog phone attached to a Digium USB fxs device. I would like the analog phone to ring when transfers are made to it, but I don't want it to ring when a call comes in from outside, although I would like the person at that phone to be able to pick up the phone and answer the incoming call. Is that possible? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parking extension not working
Okay here is an update... With my extensions.conf and parking.conf,
etc.. Files, I was able to place a call with the usb phone (analog on
zap/3) and park it by pressing flash, then typing in 701... From the
SIP phones I am still unable to transfer to 701. What does that mean?
Is that something to do with the T instead of t?
Thanks for you patience and help through this...
Sean
-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 1:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Parking extension not working
Sean Garland wrote:
Okay, lets start over... I have basically a simple parking.conf, I
use
701 as the parkext because of Andy Powells suggestion, I use a range
of 702-710. I have in my [sip] context and include = default and in
the [default] context, I have the stock include = localcalls which
has on include = parkedcalls. When asterisk is loaded and I type
show dialplan at the console, I get a list of what appears to be all
the extensions available to me and their respective contexts. Nowhere
in that list is an extension 701. Now when a call comes in, my
Polycom phone rings and I answer it. If I want to park the call I
press the conference button, and type 701, which then gives me a busy
signal for a few seconds, and the call is terminated. I ran a sip
debug and did the same procedure, and I can see that the polycom phone
requests extension [EMAIL PROTECTED] but then there is a 404 not found
error. Am I doing something wrong in the transfer procedure? When I
dial # on the polycom phone I get an instant busy, which I assume
means that my phone doesn't like that digit.
So maybe my question should be, is anyone using polycom sip phones and
able to use the parked call feature?
Sean
-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 6:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Parking extension not working
Sean Garland wrote:
Yes, I have the include = parkedcalls in the default context which
is
where my calls start from, in fact if I try to transfer to one of
the parked locations as a test of transferring to the 7xx area, I
get the ...no parked call.. message so it seems like the context
is working.
All of the parking location extensions show up in the dialplan.
What should I see on the console when transferring to 700 (or 701 as
it may
be) from SIP? Will it work from sip? Is there a way I can test
from the console?
Thanks
Sean
I don't think you can transfer to a park extension directly. In other
words, you transfer to the parking lot ( default exten 700 ) and
Asterisk parks the call in whichever parking space is available. Then
Asterisk reads back the number of the parking space. On my system,
with 700 as the park extension and 701-720 as parking spaces, I get
the message Sorry, there is no call parked on that extension if I
try to transfer to 701 which is what I would expect since it's only
exten 700 which has the special park functionality.
-Lance
ok, let's try again
In extensions.conf, do you have either a t or T as an option on your
dial string like this:
exten = s,203,Dial(${ARG2},20,rtT) ; Ring the interface, 20 seconds
maximum
If not, transfer isn't going to work and therefor park wont work either.
-Lance Arbuckle
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[Asterisk-Users] Caller id and callback
Is there a way to append a 91 in front of all incoming caller id numbers? What I am interested in is this - when a call comes in and the caller ID comes across, it is in the format 9165551212. That would be fine, but if I want to call them right back, and I choose that call and 'dial' on my phone, it fails because it hasn't dialed a 9 first. I assume also that if there was a 9, it would still fail because SBC hasn't figured out how to automatically add the 1 (like my cell phone does). Thanks Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parking extension not working
I have just set the parking extension at 701 and then the range is 702-710 and still I cannot transfer to 701. Show Dialplan doesn't show an extension 700, although it shows all the parked location extensions. If I transfer to 702, I get the message telling me that there is no parked call there Still lost!!! Thanks Sean -Original Message- From: Girish Gopinath [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 11:07 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Parking extension not working From Andy Powells Getting Started With Asterisk (V 0.1a) http://www.automated.it/guidetoasterisk.htm parking.conf file has this number set at 700. I've changed mine to 701 because I was having an issue with Asterisk - although it would 'see' (looking at the console) I had tried to transfer to 700 it appeared not to believe that I had dialed it. This was essentially due to the 00 in the 700, changing it to 701 eliminates the problem completely. Hope it helps... Girish From: Sean Garland [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Parking extension not working Date: Tue, 13 Jan 2004 16:07:56 -0800 I have the standard parking.conf but extension 700 doesn't show up in my dialplan Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos = 701-705 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 300 ; Number of seconds a call can be parked f *CLI Show dialplan [ Context 'parkedcalls' created by 'res_parking' ] '701' = 1. ParkedCall(701) [res_parking] '702' = 1. ParkedCall(702) [res_parking] '703' = 1. ParkedCall(703) [res_parking] '704' = 1. ParkedCall(704) [res_parking] '705' = 1. ParkedCall(705) [res_parking] Sean Garland Siskiyou Technology Consultants ___ 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 _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ 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] Parking extension not working
Yes, I have the include = parkedcalls in the default context which is where my calls start from, in fact if I try to transfer to one of the parked locations as a test of transferring to the 7xx area, I get the ...no parked call.. message so it seems like the context is working. All of the parking location extensions show up in the dialplan. What should I see on the console when transferring to 700 (or 701 as it may be) from SIP? Will it work from sip? Is there a way I can test from the console? Thanks Sean -Original Message- From: Lance Arbuckle [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 10:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Parking extension not working Sean Garland wrote: I have just set the parking extension at 701 and then the range is 702-710 and still I cannot transfer to 701. Show Dialplan doesn't show an extension 700, although it shows all the parked location extensions. If I transfer to 702, I get the message telling me that there is no parked call there Still lost!!! Thanks Sean Did you put include = parkedcalls somewhere in your extensions.conf that your extension phones can get to it ? -Lance ___ 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] Parking extension not working
Yeah, I did a 'restart now'... BTW, none of my responses are even making it to the list, at least not the same day as my post. Whats up with that? Sean -Original Message- From: Walt Reed [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 12:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Parking extension not working On Wed, Jan 14, 2004 at 10:36:54AM -0700, Jared Smith said: On Wed, 2004-01-14 at 09:15, Sean Garland wrote: I have just set the parking extension at 701 and then the range is 702-710 and still I cannot transfer to 701. Show Dialplan doesn't show an extension 700, although it shows all the parked location extensions. If I transfer to 702, I get the message telling me that there is no parked call there Still lost!!! Did you restart Asterisk after making the changes to parking.conf? Sounds like maybe you forgot to... BTW, A simple reload is NOT enough It does need to be a full restart. ___ 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
[Asterisk-Users] Parking extension not working
I have the standard parking.conf but extension 700 doesn't show up in my dialplan Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos = 701-705 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 300 ; Number of seconds a call can be parked f *CLI Show dialplan [ Context 'parkedcalls' created by 'res_parking' ] '701' = 1. ParkedCall(701) [res_parking] '702' = 1. ParkedCall(702) [res_parking] '703' = 1. ParkedCall(703) [res_parking] '704' = 1. ParkedCall(704) [res_parking] '705' = 1. ParkedCall(705) [res_parking] Sean Garland Siskiyou Technology Consultants ___ 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] call parking
I am having trouble with call parking I am basically using the stock sample files, but extension 700 doesnt show up in my dialplan. When I transfer a call to 700, I get the fast busy like there is extension 700 HELP! Sean Garland
RE: [Asterisk-Users] Multi-line help
Thank you all for your responses. Since I was a phone installer (previous life) and installed Lucent Partner and Merlin systems, I was on the key system mode of thinking. On the Polycom phones each line button is a registration, so I wonder how I could program a SIP registration to speed dial a number? Would that be done through exten.conf like: [button2] Exten = 1,dial(zap/g1/5551212) ??? So then, carrying over to key system terms, I would basically be setting up line pool buttons... Basically with my small office (2 phones, and one * box with 2 x100p cards) I would just use the first button (or whatever) for my registration with my * and call it good... I am thinking of proposing this system to my partner corp which would entail around 13 extensions and 6 lines... How would I give someone upstairs the ability to view if each user was on the phone or not? -- should probably be a new thread Currently they have 18 button phones that are programmed with the incomming lines, then the users (LED's glow when user is on). This is so much fun! (no really!) Sean -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Monday, January 05, 2004 6:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Multi-line help Sean, Basically I guess I am thinking of the traditional key systems approach which is to have the CO lines appear on the phone. The problem it appears with SIP (not really *) and the particular phones, is to have the reporting. I guess what I was looking for was to have the buttons not only represent the incoming lines, but to also show their status (busy, hold, etc...). As you've already mentioned, what you've described is a key system, and not a pbx. (There might be an open source key system out there somewhere.) On that note, what (typically with SIP/*) are the multiple line phone buttons used for? I know you have to have at least one for access to the asterisk system, but what is the point of the multiple registrations? Several reasons depending upon the actual requirements... 1. Small office, Customer Service appears on line 1, President on line 2. Answering line 2 with an appropriate messages (when he's not around) is different then answering line 1 as a Customer Service person. 2. Shared tenant service: five different businesses in the same small complex. The receptionist has all five lines on her phone, and answers with an appropriate message for each business when their lines are unanswered. 3. Home-boy (no asterisk) subscribes to two different VoIP providers with two different rate plans. Line 1 registers with provider 1, and line 2 with provider 2. You choose which service you want based on your knowledge of what your trying to accomplish (not necessarily programmable if an * system was included). 4. On the Cisco 7960, I have one of the line buttons programmed as a speed dial to a certain extn as I'm calling it often. 5. Remote intercom: place a speaker phone by the front door and configure it for auto answer. When the doorbell rings, push one of your preprogrammed buttons to speak to who's at the door. 6. You could probably program * to open the garage door with one of the buttons. ;) As Steve pointed out earlier today, there are many ways to accompish the same function within asterisk, therefore some of the items listed above might be done a different way. That's fine. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Message waiting indicator
Thanks, the phones that I have Polycom Soundpoint IP 500's. In the specific config file for the phone itself, there are some lines that have to do with MWI and there are three settings to set. Here is the section of the manual for the phone msg.mwi.x.subscribe ASCII encoded string containing If non-Null, the telephone digits (the user part of a SIP will send a URL) or a string that constitutes SUBSCRIBE request a valid SIP URL (6416 orto this contact after [EMAIL PROTECTED]) boot-up. msg.mwi.x.callBackMode contact or registration If set to contact, a call will be placed to the contact specified in the callback attribute when the user invokes message retrieval. If set to registration, a call will be placed using this registration to the contact registered (the telephone will call itself). msg.mwi.x.callBack ASCII encoded string containing Contact to call when digits (the user part of a SIP retrieving messages URL) or a string that constitutes for this registration. a valid SIP URL (6416 or [EMAIL PROTECTED]) Does this mean that if the sip entry comes out to [EMAIL PROTECTED], is that what I put in for the subscribe and callback? I don't understand the connection between the SUBSCRIBE feature and the NOTIFY Anyone with Polycom experience with MWI? I will have to check to see if the NOTIFY is even happening... Sean -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 6:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Message waiting indicator What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? I think the examples that you might have looked are suggesting that when a voicemail is left for a single extension, you can place definitions in your sip.conf file that turn on the MWI (message waiting indicator) LED on more then one phone. (I'll leave that up to you to figure out whether that is a feature of use to you.) Asterisk will occasionally look in the /var/spool/asterisk/voicemail/default/3008/INBOX directory (where 3008 represents the extension number), and if a certain file exists, send a sip message to the extn(s) that you defined in sip.conf as mailbox=3008. The sip message sent to the phone (in hex using a packet sniffer) looks like: 0020: c1 5b 13 c4 13 c4 01 e2 58 97 4e 4f 54 49 46 59 | Á[.Ä.Ä.âX-NOTIFY 0030: 20 73 69 70 3a 33 30 30 38 40 32 30 35 2e 32 31 | sip:[EMAIL PROTECTED] 0040: 32 2e 31 39 33 2e 37 31 20 53 49 50 2f 32 2e 30 | 2.173.91 SIP/2.0 0050: 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 | ..Via: SIP/2.0/U 0060: 44 50 20 32 30 35 2e 32 31 32 2e 31 39 33 2e 31 | DP 205.212.193.1 0070: 30 31 3a 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a | 01:5060;branch=z 0080: 39 68 47 34 62 4b 33 63 31 63 61 35 65 31 0d 0a | 9hG4bK3c1ca5e1.. 0090: 46 72 6f 6d 3a 20 22 61 73 74 65 72 69 73 6b 22 | From: asterisk 00a0: 20 3c 73 69 70 3a 61 73 74 65 72 69 73 6b 40 32 | sip:[EMAIL PROTECTED] 00b0: 30 35 2e 32 31 32 2e 31 39 33 2e 31 30 31 3e 3b | 05.212.193.101; 00c0: 74 61 67 3d 61 73 35 37 63 63 64 33 32 65 0d 0a | tag=as57ccd32e.. 00d0: 54 6f 3a 20 3c 73 69 70 3a 33 30 30 38 40 32 30 | To: sip:[EMAIL PROTECTED] 00e0: 35 2e 32 31 32 2e 31 39 33 2e 39 31 3e 0d 0a 43 | 5.212.193.91..C 00f0: 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 61 73 74 | ontact: sip:ast 0100: 65 72 69 73 6b 40 32 30 35 2e 32 31
RE: [Asterisk-Users] Echo with polycom phones
Yup, two x100p cards. The echo goes away really quickly, but the first second or two is pretty echoey (is that a word?) Sean -Original Message- From: Christian Hecimovic [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 10:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo with polycom phones Hey Sean, Are you using Zap modem cards (X100P)? There can be bad echo with those things. The echo canceling effect you are hearing comes from the Polycom phones - they dynamically learn from the echo at the beginning of the call and adjust the echo cancellation accordingly. We are using a Mediatrix SIP gateway now - no echo at all. Christian On Monday 05 January 2004 13:28, Sean Garland wrote: I have soundpoing ip 500 phones and the first few seconds of every call has echo, which then goes away. Is there a way to have the echo cancel on at the beginning? It seems like it is testing at the beginning but it would be nice if I could have it start closer Thanks Sean Garland Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo with polycom phones
Title: Echo with polycom phones I have soundpoing ip 500 phones and the first few seconds of every call has echo, which then goes away. Is there a way to have the echo cancel on at the beginning? It seems like it is testing at the beginning but it would be nice if I could have it start closer. Thanks Sean Garland Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED]
RE: [Asterisk-Users] Multi-line help
Basically I guess I am thinking of the traditional key systems approach which is to have the CO lines appear on the phone. The problem it appears with SIP (not really *) and the particular phones, is to have the reporting. I guess what I was looking for was to have the buttons not only represent the incoming lines, but to also show their status (busy, hold, etc...). On that note, what (typically with SIP/*) are the multiple line phone buttons used for? I know you have to have at least one for access to the asterisk system, but what is the point of the multiple registrations? Thank you to all. Sean Garland Siskiyou Technology Consultants -Original Message- From: Nicholas Comanos [mailto:[EMAIL PROTECTED] Sent: Sunday, January 04, 2004 8:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Multi-line help Could you explain in a little more detail about what you are trying to do with the multi-lines? Maybe a more in depth example would help. In my (limited) experience, I have seen two types of multi-line uses 1. The phone has a number of lines (usually) two. If the first line is busy, the call rings on the second and so the user has the option of putting the first on hold and answering the new incoming call or letting it ring out. Normally the user has only one advertised extension number (and the second line may not even have its own unique extension #). The second line is often used for inquiry calls or if the primary line is busy. Usually the phone selects the first available line when making a new call. 2. The second type of multi-line use I have seen is where one phone has lines for multiple extensions and those extensions may be represented on multiple phones (shared line). For example, the phone of a personal assistant may have a line for them and their boss. The multi-line button in this case may often shows the status (ie: busy) of the extension as it is 'shared' among multiple phones. Depending on the configuration, if the extension is called, it may ring on one or more of the phone lines that support that extension. Even in this case, the phone often has a default line to use when the handset is picked-up to make a call. Asterisk will support the type 1. above as long as the handset support multiple lines (which in your case it does,) However, case 2, I do not believe is supported by Asterisk at the moment - you can make the line ring, but you will not be able to show the status of the line on other phones. In addition, with a SIP phone, the phone will also have to have a way of receiving the notification for the status of the line (busy, not busy.) - not all SIP phones support this, but looking through the Polycomm manual, it seems they do. - Original Message - From: Sean Garland [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 04, 2004 7:18 PM Subject: [Asterisk-Users] Multi-line help I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. Thanks a ton. I have been testing asterisk and on the mailing list for about a month now... I would be happy to send all my config files for perusal. Sean Garland - Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This is a test
Title: This is a test It appears that my replies aren't getting to the list. Just testing to see what is going on Sean
[Asterisk-Users] Echo on polycom sip phone
I have soundpoing ip 500 phones and the first few seconds of every call has echo, which then goes away. Is there a way to have the echo cancel on at the beginning? It seems like it is testing at the beginning but it would be nice if I could have it start closer Thanks Sean Garland Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message waiting indicator
What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? Thanks again. Just some background as you start seeing my lists, I just started my own business this year (open 1 day officially so far) and I chose, back in November/December to use * as my phone system. Currently I am using it as my main phone system, and have voicemail configured, with two lines, and two Polycom phones. I hope to post my entire experience on the wiki or something when I am satisfied that I am far enough along... Great product, but the docs need some work. I would be interested in helping with that when the time comes. Sean Garland Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi-line help
Title: RE: [Asterisk-Users] Multi-line help *** I am resending this because something is wrong with the reply function *** Basically I guess I am thinking of the traditional key systems approach which is to have the CO lines appear on the phone. The problem it appears with SIP (not really *) and the particular phones, is to have the reporting. I guess what I was looking for was to have the buttons not only represent the incoming lines, but to also show their status (busy, hold, etc...). On that note, what (typically with SIP/*) are the multiple line phone buttons used for? I know you have to have at least one for access to the asterisk system, but what is the point of the multiple registrations? Thank you to all. Sean Garland Siskiyou Technology Consultants -Original Message- From: Nicholas Comanos [mailto:[EMAIL PROTECTED]] Sent: Sunday, January 04, 2004 8:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Multi-line help Could you explain in a little more detail about what you are trying to do with the multi-lines? Maybe a more in depth example would help. In my (limited) experience, I have seen two types of multi-line uses 1. The phone has a number of lines (usually) two. If the first line is busy, the call rings on the second and so the user has the option of putting the first on hold and answering the new incoming call or letting it ring out. Normally the user has only one advertised extension number (and the second line may not even have its own unique extension #). The second line is often used for inquiry calls or if the primary line is busy. Usually the phone selects the first available line when making a new call. 2. The second type of multi-line use I have seen is where one phone has lines for multiple extensions and those extensions may be represented on multiple phones (shared line). For example, the phone of a personal assistant may have a line for them and their boss. The multi-line button in this case may often shows the status (ie: busy) of the extension as it is 'shared' among multiple phones. Depending on the configuration, if the extension is called, it may ring on one or more of the phone lines that support that extension. Even in this case, the phone often has a default line to use when the handset is picked-up to make a call. Asterisk will support the type 1. above as long as the handset support multiple lines (which in your case it does,) However, case 2, I do not believe is supported by Asterisk at the moment - you can make the line ring, but you will not be able to show the status of the line on other phones. In addition, with a SIP phone, the phone will also have to have a way of receiving the notification for the status of the line (busy, not busy.) - not all SIP phones support this, but looking through the Polycomm manual, it seems they do. - Original Message - From: Sean Garland [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 04, 2004 7:18 PM Subject: [Asterisk-Users] Multi-line help I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. Thanks a ton. I have been testing asterisk and on the mailing list for about a month now... I would be happy to send all my config files for perusal. Sean Garland - Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-line help
I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom (internal extension) button. I have managed to get the line 1 to ring on the line 1 button and the same for line two. I have even managed to get extension transfers to happen on the itcm button. The trouble I have is that I don't know if someone else is on the particular line, and when I dial, it picks up the first available button (line) so even if I dial an extension, it looks like I am dialing from line 1 to the extension. How do I make it pick the third button, etc... Confusing? I have read the handbook and countless searches through wiki and Google, but cannot find practical examples of multi-line use with asterisk. Thanks a ton. I have been testing asterisk and on the mailing list for about a month now... I would be happy to send all my config files for perusal. Sean Garland - Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Sip Registration
Hey, I am currently working on a Polycom 500 phone Asterisk solution, and the key is definitely to use the xml config files that Matt spoke of. That combined with an FTP server (setup like the sip docs say) work very well in getting the phone to do what you want. It then becomes getting the config files for Asterisk that will make it all work. I will update on what I finally have when I am done Sean -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 9:02 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Polycom Sip Registration Hello, The best thing to do is to use the XML config files. the web interface isn't the best way to do anything, it's best to kind of ignore it. MATT--- -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Friday, December 26, 2003 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom Sip Registration Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-line, multi-registration phones
Title: Multi-line, multi-registration phones I have hard phones that are capable of handling three calls at once. That is setup (apparently) through multiple registrations. My question is has anyone done this and what is the proper way of doing it? Do I have to setup (for 2 phones that have three lines) 6 sections in my sip.conf and setup 6 extensions to handle the registrations? Also, if I found by searching the web sample code for making both sip extensions ring when a call comes in, but what if I had 100 extensions? Seems like the string would get pretty long, is there a way to put all extensions in a single group and ring the group? All kinda is the same question. But thanks for the answer anyway Sean Garland
[Asterisk-Users] X100p always busy - update
Title: X100p always busy - update Well, after bummin around thinking I had bad fxo cards, I finally discovered that I was loosing two phones in my home when I had the cards plugged in Turns out the jack I had plugged into, was wired for two phones (line 1, green/red and line 2, yel/blk) and I also was using a 4 wire phone cord for the connection. It turns out that the x100p cards are wired in such a way internally that if you are running a 4 wire phone cord to them, that you might short out between the two lines at the jack (which worked separately with phones and computers) and get funky results.. Moral of the story is to always use two wire phone cords with the x100p fxo cards. Problem solved, and I was able to continue my development. Thanks Sean Garland
[Asterisk-Users] X100p problem
Title: X100p problem I am having a problem with the x100p cards. It doesn't matter whether the card is in the machine or not, all I get is a busy signal when calling. The Asterisk box doesn't give me any errors and doesn't show that any call is coming through. I removed the cards from the machine completely and they still give busy signal when dialed. Any ideas? I must say that after dealing with the ordering process with Digium, and now the seemingly broken cards, I have to say that I completely frustrated and unhappy with deciding to go with digium. I think that Asterisk is probably very cool, and will do what I want, but it took three weeks to get my cards and the people at digium won't email to save their lives. Anyway, please help with the card problem as I feel that I am out another week and this was supposed to be running last week Thanks Sean Garland
[Asterisk-Users] Polycom SIP Phone config files
I have read on this list that the config files might be available to make them work on Asterisk? If that is so, could someone please email them to me? We have the Polycom Soundpoint IP 500 phones. Thanks a bunch, my goal is to make this phone and asterisk my business system. Thanks Sean [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
