Re: [asterisk-users] Sip phone does not call
The two phones belong to context phones and the two extensions are in context internal. In context phones you need to include => internal so that context phones knows about those extensions. Or put the two extensions in context phones and not context internal. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 19, 2010, at 2:05 PM, ayodele abejide wrote: > Hello group, > > I have asterisk running on my ubuntu machine, and I have a peer to peer > network with an XP machine, both of the running x-lite client, I try calling > either of the soft phone from the other and the response I get is on my > asterisk console is as below: > > > [May 19 19:31:18] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call > from '1000' to extension '3000' rejected because extension not found. > > [May 19 19:31:39] NOTICE[1476]: chan_sip.c:21298 handle_request_subscribe: > Received SIP subscribe for peer without mailbox: 1000 > > [May 19 19:31:44] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call > from '1000' to extension '1000' rejected because extension not found. > > > My Diaplan Settings (extensions.conf) > > [globals] > > > [general] > autofallthrough=yes > > > [default] > exten => s,1,Verbose(1|Unrouted call handler) > exten => s,n,Answer() > exten => s,n,Wait(1) > exten => s,n,Playback(tt-weasels) > exten => s,n,Hangup() > > > [incoming_calls] > > > [internal] > exten => 1000,1,Verbose(1|Extension 1000) > exten => 1000,n,Dial(SIP/1000,30) > exten => 1000,n,Hangup() > > > exten => 3000,1,Verbose(1|Extension 3000) > exten => 3000,n,Dial(SIP/1000,30) > exten => 3000,n,Hangup() > > > Sip Settings (sip.conf) > > [general] > context=default > bindport=5060 > srvlookup=yes > > [1000] > type=friend > host=dynamic > context=phones > > [3000] > type=friend > host=dynamic > context=phones > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. > -- > _ > -- 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 -- _ -- 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] SIP Phone for conference room use.
On Thu, 11 Mar 2010 12:56:05 +0100, Tommy Botten Jensen wrote: >-BEGIN PGP SIGNED MESSAGE- >Hash: SHA512 > >Hi > >I'm looking for a good phone SIP phone for conference room use. > >My requirements are in order: >* Speaker quality >* External microphone support. >* Provisioning support / asterisk compatibility. > >Does anyone have any experience on this field? > >I'm willing to stretch on price to get decent quality. Any Polycom Soundstation model will be more than acceptable. We use the IP6000 and love it. Meets your criteria for extension mic's & speakers. The IP4000 model was discontinued, and did not support HDVoice. HDVoice makes a huge difference if you can make calls IP-to-IP. It makes conferencing seriously better. I've also trialed the Konftel/snom MeetingPoint (http://www.mgraves.org/voip/2009/07/initial-impressions-konftel-300-spe akerphone/) . It's very good as well, and can be had with a diversity of interfaces; analog, IP & USB making it very versatile. It also records to a SD card if you need. Michael Graves -- Michael Graves mgravesmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] SIP Phone for conference room use.
On 11 Mar 2010, at 12:43, Gordon Henderson wrote: > On Thu, 11 Mar 2010, Tommy Botten Jensen wrote: > >> Hi >> >> I'm looking for a good phone SIP phone for conference room use. >> >> My requirements are in order: >> * Speaker quality >> * External microphone support. >> * Provisioning support / asterisk compatibility. >> >> Does anyone have any experience on this field? >> >> I'm willing to stretch on price to get decent quality. > > Polycom IP4000 with external mics. > > Expensive, but they've been doing conference phones for a very long time > so seem to know what they're doing. We've just got some Snom MeetingPoints and are pleased with them so far.. W -- _ -- 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] SIP Phone for conference room use.
On Thu, 11 Mar 2010, Tommy Botten Jensen wrote: > Hi > > I'm looking for a good phone SIP phone for conference room use. > > My requirements are in order: > * Speaker quality > * External microphone support. > * Provisioning support / asterisk compatibility. > > Does anyone have any experience on this field? > > I'm willing to stretch on price to get decent quality. Polycom IP4000 with external mics. Expensive, but they've been doing conference phones for a very long time so seem to know what they're doing. The few I've installed, I've just used their web interface to setup the SIP account. Royal PITA as it seemed to require a reboot for every change, however you get there in the end... Gordon -- _ -- 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] SIP phone recommendation (used to be: no subject)
I'd go with Polycom all the way. We have a number of different types of phones in use, or that we've worked with, including Grandstream, SIpura and Atacom, and the quality difference with the Polycom phones is astounding. On 10/29/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > My apologies to the list for not having entered a subject line in the > email. > > Thanks > > On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: > > > Hi all, > > > > We have a client that needs to setup about 80 desk phones (about 50 > > in one location and about another 30 in 5 different locations). Which > > brand/model would you recommend. We were personally thinking in > > recommending either Cisco, Aastra, Polycom, or Snom, for we've heard > > great things about them. However, having no real experience with them > > makes it hard in recommending one to our customer. The only > > experience we've had is a very frustrating one trying to load the IP > > software on a Cisco 7970G and so we assume that if we have to go > > through that for all 80 phones, we'll probably commit suicide :) > > > > Thanks > > > > > > ___ > > --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 > > > ___ > --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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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] SIP phone recommendation (used to be: no subject)
Michael Graves wrote:
> On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote:
>
>> Well, just general office use. They are a real-state construction
>> company, so the phones will get some heavy use since most of the
>> phones are going to sales associates.
>>
>> Now, one of the things we are most interested in are:
>> 1) Asterisk compatibility
>> 2) Mass provisioning
>> 3) Remote management
>> 4) Excellent audio quality (I know there are many factors involved,
>> but would like to rule out the phone set itself)
>> 5) Robustness
>> 6) Vendor reputation and warranties
>>
>> We have used Linksys 941s in the past and think they're pretty good.
>> However, we've only used them in 3-5 phones office environments.
>> We've also used the Polycoms IP 501 and 650s. They seem good, but
>> sometimes the users complain about the audio being a bit weird in the
>> sense that, probably, the silence detection may give the user a
>> feeling that the line dropped. Then again, we've only used these once
>> (one client installation for each), so for practical purposes, we
>> don't really have any larger quantity real-life experience.
>
> For my money it's Polycom every time. It's great hardware. Meets all
> your requirements.
Granted I have only used Polycom phones, but I would second that vote.
My experience with provisioning is that it isn't necessarily hard but it
can be time consuming. Your best bet is to get your firmware extracted
and then go through the sip.cfg line by line with the admin guide handy
and tweak as you go. Then repeat with the default phone.cfg file. I use
a shell script (which I'll share with anyone who wants it) that makes
adding additional phones a snap. I pass it the name of the default
template file, the extension number and the MAC address of the phone and
it creates the MAC.cfg and phone{extension}.cfg files.
> I thought that silence supression was specifically disallowed with
> Asterisk? Something about timing requirements not being met.
I can't say for certain, but that may not be true any more. I came
across a setting called "internal_timing" that may allow for the use of
silence suppression. If anyone can comment on that I'd be interested to
hear what that setting does. This is what I found from Google:
http://forums.digium.com/viewtopic.php?t=15577
http://bugs.digium.com/view.php?id=5374
-Dave
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Re: [asterisk-users] SIP phone recommendation (used to be: no subject)
On Mon, 29 Oct 2007 15:01:38 -0400, [EMAIL PROTECTED] wrote: >Well, just general office use. They are a real-state construction >company, so the phones will get some heavy use since most of the >phones are going to sales associates. > >Now, one of the things we are most interested in are: >1) Asterisk compatibility >2) Mass provisioning >3) Remote management >4) Excellent audio quality (I know there are many factors involved, >but would like to rule out the phone set itself) >5) Robustness >6) Vendor reputation and warranties > >We have used Linksys 941s in the past and think they're pretty good. >However, we've only used them in 3-5 phones office environments. >We've also used the Polycoms IP 501 and 650s. They seem good, but >sometimes the users complain about the audio being a bit weird in the >sense that, probably, the silence detection may give the user a >feeling that the line dropped. Then again, we've only used these once >(one client installation for each), so for practical purposes, we >don't really have any larger quantity real-life experience. For my money it's Polycom every time. It's great hardware. Meets all your requirements. I thought that silence supression was specifically disallowed with Asterisk? Something about timing requirements not being met. Michael -- Michael Graves mgravesmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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] SIP phone recommendation (used to be: no subject)
Well, just general office use. They are a real-state construction company, so the phones will get some heavy use since most of the phones are going to sales associates. Now, one of the things we are most interested in are: 1) Asterisk compatibility 2) Mass provisioning 3) Remote management 4) Excellent audio quality (I know there are many factors involved, but would like to rule out the phone set itself) 5) Robustness 6) Vendor reputation and warranties We have used Linksys 941s in the past and think they're pretty good. However, we've only used them in 3-5 phones office environments. We've also used the Polycoms IP 501 and 650s. They seem good, but sometimes the users complain about the audio being a bit weird in the sense that, probably, the silence detection may give the user a feeling that the line dropped. Then again, we've only used these once (one client installation for each), so for practical purposes, we don't really have any larger quantity real-life experience. Thanks On Oct 29, 2007, at 2:18 PM, Eric Chamberlain wrote: > What is the use case? > > Linksys, Polycom, Snom, and Aastra all have their strengths and > weaknesses. > > -- > Eric Chamberlain, CISSP > Chief Technical Officer > Voxilla - http://voxilla.com/ > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] >> Sent: Monday, October 29, 2007 10:42 AM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] (no subject) >> >> Hi all, >> >> We have a client that needs to setup about 80 desk phones (about 50 >> in one location and about another 30 in 5 different locations). Which >> brand/model would you recommend. We were personally thinking in >> recommending either Cisco, Aastra, Polycom, or Snom, for we've heard >> great things about them. However, having no real experience with them >> makes it hard in recommending one to our customer. The only >> experience we've had is a very frustrating one trying to load the IP >> software on a Cisco 7970G and so we assume that if we have to go >> through that for all 80 phones, we'll probably commit suicide :) >> >> Thanks >> >> >> ___ >> --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 > > > > > ___ > --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 ___ --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] SIP phone recommendation (used to be: no subject)
My apologies to the list for not having entered a subject line in the email. Thanks On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: > Hi all, > > We have a client that needs to setup about 80 desk phones (about 50 > in one location and about another 30 in 5 different locations). Which > brand/model would you recommend. We were personally thinking in > recommending either Cisco, Aastra, Polycom, or Snom, for we've heard > great things about them. However, having no real experience with them > makes it hard in recommending one to our customer. The only > experience we've had is a very frustrating one trying to load the IP > software on a Cisco 7970G and so we assume that if we have to go > through that for all 80 phones, we'll probably commit suicide :) > > Thanks > > > ___ > --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 ___ --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] SIP phone supporting more than 10 extension with acall transfer command
Try the Snom 360. The softphone version of it (a free demo) has 12 lines (I presume the real thing has the same). You can find the softphone at http://www.snom.com/download/snom360-5.3.exe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of younss azzayani Sent: Friday, March 16, 2007 5:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP phone supporting more than 10 extension with acall transfer command Hi every body, can someone please tell me about a SIP phone that support more than 10 extension (free or not free ;) ) wich will be used in my company, i've bought a SNOM but it just support 5 sip extension Kind regards ___ --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] SIP phone supporting more than 10 extension with a call transfer command
ok thank you :) i'll look for this ___ --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] SIP phone supporting more than 10 extension with a call transfer command
snom320, snom360 and snom370 are supporting 12 different SIP identities. Regards, Sven On Friday 16 March 2007 10:57, younss azzayani wrote: > Hi every body, > can someone please tell me about a SIP phone that support more than 10 > extension (free or not free ;) ) wich will be used in my company, i've > bought a SNOM but it just support 5 sip extension > Kind regards > ___ > --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 -- --- See our Docs, FAQs, etc at: http://snom.com/wiki --- Sitz/Domicile: snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.com Handelsregistereintrag/Register of Corporations entry: Berlin-Charlottenburg HRB 612842 Vorstand/Executive Board: Dr. Christian Stredicke, Dr. Michael Knieling, Alexander Khan Vorsitzender des Aufsichtsrates/Chairman of the Supervisory Board: Prof. Dr. Harald Melcher --- ___ --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] Sip Phone CID
try actually setting the rpid in the dialplan using sipcalledrpid(name,number) Rob Schall wrote: I set both the trustrpid and sendrpid to "yes", but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say "1001", it should say "Jim <1001>". Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --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] Sip Phone CID
I set both the trustrpid and sendrpid to "yes", but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: > check out rpid > > Mark Johnson wrote: >> >> >> Rob Schall wrote: >>> This might sound like an odd question but here it is anyways... >>> >>> We currently have Polycom 501 phones. We have Asterisk with >>> Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone >>> dials another, the receiving end does in fact see the callers ID. >>> But... >>> our old phone system set the caller id on the senders phone to show who >>> they called. >>> >>> Example... >>> >>> If Sally calls Jim, then Sally's phone should just say "1001", it >>> should >>> say "Jim <1001>". >>> >>> >>> Any know if this is possible. Our old PBX did this, and the bosses were >>> curious if this is possible. >>> >>> Thanks, >>> Rob >>> >>> >> I have tried over and over to figure out how to do this and it >> doesn't seem possible at the moment. I know this can be done with >> chan_sccp and maybe even chan_skinny (haven't tried that in a few >> years), but you'd need Cisco phones to do it. Is this something on >> anyone's To-Do list? >> >> Thanks, >> >> Mark >> ___ >> --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] Sip Phone CID
check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say "1001", it should say "Jim <1001>". Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --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] Sip Phone CID
Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say "1001", it should say "Jim <1001>". Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --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] sip phone networking question [possibly OT]
> I was wondering if we could uplink small switches to the wall data ports to > the switch, and connect the additional SIP phones to them to get them > connectivity to Asterisk? Yes, we do it and it works fine, as long as you don't cascade more than 3 switches between two devices your latency should be fine, also make sure they are good switches like a 3com and not a crappy dlink. ___ --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] sip phone networking question [possibly OT]
I have some ethernet cable splitters I'm not using any more. They go in pairs, one plugs into the wall socket in the office, the other plugs into the other end of the same cable in the server room. each gives two female ethernet sockets that represent two separate network cables, each using two of the pairs. Works great at 100Mb speeds, we needed one of the lines to be gigabit and they weren't cutting it for that so we ran more cables. This might be considerably cheaper than extra switches, but your main switch has to have enough ports to accommodate the extra stations. Contact me off-list if you would like to purchase them. I have 9 pairs, which would turn 9 cables into 18. Moj T. Shaw wrote: I have a client that is looking for a "least cost" solution of providing more SIP phones to an existing asterisk setup. The Issue is this: He has 7 total data run lines running back to the switch/phone room (small company). However the want to add a total of 5 or more Phones. He doesn't want to foot the cost of running more data lines along the middle of a large room all the way to the walls and then back to the phone room. I was wondering if we could uplink small switches to the wall data ports to the switch, and connect the additional SIP phones to them to get them connectivity to Asterisk? As long as each phone has seperate IP, i'm not sure if there would be a problem. Anyone care to chime in to point out any pitfalls or potential problems with this setup? Thanks! Terrelle ___ --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 !DSPAM:500,44ce77bb139581298614243! -- Mojo <[EMAIL PROTECTED]> Office Manager, Horan & Company, LLC (907) 747- x112 ___ --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] sip phone networking question [possibly OT]
Should be no problems at all. But keep in mind that you're getting what you pay for. There have been a few posts recently regarding issues with a Polycom 501 that turned out to be (partially) related to the networking equipment used. Make sure you get a good quality switch (and NOT a hub, as Brandon's pointed out), and you'll be fine. Alex On 7/31/06, T. Shaw <[EMAIL PROTECTED]> wrote: I have a client that is looking for a "least cost" solution of providingmore SIP phones to an existing asterisk setup. The Issue is this: He has 7 total data run lines running back to theswitch/phone room (small company).However the want to add a total of 5 or more Phones. He doesn't want to footthe cost of running more data lines along the middle of a large room all the way to the walls and then back to the phone room.I was wondering if we could uplink small switches to the wall data ports tothe switch, and connect the additional SIP phones to them to get themconnectivity to Asterisk? As long as each phone has seperate IP, i'm not sure if there would be aproblem.Anyone care to chime in to point out any pitfalls or potential problems withthis setup?Thanks! Terrelle___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___ --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] sip phone networking question [possibly OT]
Shouldn't be a problem as long as you're using switches and not hubs, and the network is atleast 100Mb.-brandonOn 7/31/06, T. Shaw < [EMAIL PROTECTED]> wrote:I have a client that is looking for a "least cost" solution of providing more SIP phones to an existing asterisk setup.The Issue is this: He has 7 total data run lines running back to theswitch/phone room (small company).However the want to add a total of 5 or more Phones. He doesn't want to foot the cost of running more data lines along the middle of a large room all theway to the walls and then back to the phone room.I was wondering if we could uplink small switches to the wall data ports tothe switch, and connect the additional SIP phones to them to get them connectivity to Asterisk?As long as each phone has seperate IP, i'm not sure if there would be aproblem.Anyone care to chime in to point out any pitfalls or potential problems withthis setup? Thanks!Terrelle___--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 -- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --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] Sip phone settings set when user registers
2006/7/27, Nik Engel <[EMAIL PROTECTED]>: User logs into any phone and the settings of the phone are always thesame. Meaning individual keyassignement is always the same.Hi,Do you mean :1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance) for unidentifed users ? I'm not sure many phones offer special keys for login-logout. 2. What should happen when users change phones settings ? Shall these changes be saved somehow (during logoffs ?) and somewhere for latter reuse ? That implies phone config should be portable from one phone to another. That doesn't seem easy if phones are installed in different locations. Regards ___ --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] Sip phone settings set when user registers
On 7/27/06, Jon Schøpzinsky <[EMAIL PROTECTED]> wrote: Hello Just use Snom or grandstream phones. They can be provisioned very easily via HTTP. You just setup a config URL on the phones, and they get their configurations from there. If you want to get more advanced, they can send along their MAC address, and thereby enabling you to custom config them directly from a central application, based on the phones MAC address. The snom phones can even be instructed to download a configuration from a URL via DHCP. This is true, but making a change to the configuration (pushing a change) based on a user action is much harder, or even impossible. We have compromised by having the phones configured once at boot, and having Asterisk change the behaviour "under the hood" when the user requests it. Not tidy, but the end result works. Cheers, Steve ___ --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] Sip phone settings set when user registers
Hi ! Also, the GXP-2000 is a very popular model too, although once you consider the capabilities of Asterisk the only real advantage this unit has over the others (even in an office environment), is the Power over Ethernet (PoE) feature: which is supported be Snoom as well. Anyway I would be more interested in a method to configure the key assignements upon login with asterisk ?? any ideas how to do that ? Nik ___ --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] Sip phone settings set when user registers
Hi Nik, I like the Grandstream Budge Tone 102 VoIP Phones which you can find here: http://www.voipsupply.com/product_info.php?products_id=40 and here: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-31609737728.htm Also, the GXP-2000 is a very popular model too, although once you consider the capabilities of Asterisk the only real advantage this unit has over the others (even in an office environment), is the Power over Ethernet (PoE) feature: Nik Engel wrote: Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Thanks for any advices Nik ___ --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] SIP phone receiving but not transmitting
Bill Maidment wrote: Hi I've been using Asterisk for a while now with the TDM400 and it seems to be working fine. I'm using version 1.2.2 and I've struck a problem when I added a Budge Tone 100 SIP phone to the network. The phone rings when calls come in and I can make calls but in all cases (internal or external calls) the other party cannot hear me even though I can hear them. Don't worry everyone! I dismantled the headset as suggested on the wiki and found a loose wire to the mike. They really are crappy phones. aren't they. Cheers Bill -- What's the difference between Linux and Windoze? Linux - Thousands of programmers are working *WITH*you. Windoze - Thousands of programmers are working *AGAINST* you. Web Site http://www.maidment.com.au ___ --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] Sip phone with Bluetooth - does it exist?
The ClipComm CP101B has WIFI capability, they make a bluetooth mini PCMCIA card for it. Or you can pick up a VXI BlueParrot BP200, it's about the least expensive BlueTooth headset/base station I know of that isn't junk, and it works well with just about any phone. Downside is it does not have a handset lifter, but other than that it's excellent. Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Joe Pukepail To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 18, 2006 9:24 PM Subject: [Asterisk-Users] Sip phone with Bluetooth - does it exist? Anyone know if a Sip phone with bluetooth for a wireless headset exists? If so does anyone have any recommendations? Or maybe a Wifi/Sip headset? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] SIP phone procedural question
> A lot of my customers have people who are in the office most of the time but > occasionally wish to work from home. So they may have a sip > phone which is extension 208 in the office. When they work from home they > can of course plug in a sip phone into their broadband > connection and work with that. But it would be ideal if they could be same > extension as phone in office. If they try to register as same sip > user - eg extn 208 - will it work. Then problem is phone on their desk will > still ring p***ing all their colleagues off. > > How do people deal with this sort of thing. Ideally, would want person to be > able to easily switch from office to home but use same > extension. > > Or does sip somehow deal with this? Is there a standard sip way of dealing > with this? You should be able to find multiple ways to do that on the wiki. Use keywords such as call forwarding in the search. One way to do it for x1234 (as a high level example only) is to: - assign 9234 as a call forward "control" extension - employee dials x9234 - asterisk writes a value into the db (see "show application dbput") - when a call is sent to x1234, dialplan code checks for value using dbget. - if value is set, ring at-home extension; if not set, ring at-office extension (eg, x1234). If you want to make the above a little more sophisticated, when the user dials x9234 prompt the user for which extension to forward his calls to and write that to the db. When a call arrives, the call is call-forwarded to whatever extension the user entered. Want to complicate that more, write a macro to do that for all extens. ___ 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 phone failover using DNS SRV?
> Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? Definitely we have been doing this for quite a while. > If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP > addresses of A and B are associated with the same name, and both servers have equal priority and equal >weight. We have this working great with both Polycom and Sipura devices. We have servers of different priority (i.e. - primary then failover). The name is the same. > In order to make calls through B after A goes down, do you have to wait as long as the registration > retry interval? Or can you make calls through B as soon as you pick up the phone and dial, because the > INVITE message through A fails, and the phone re-sends the INVITE through B? The way we have it working A is a higher priority than B, so every phone will register to A unless it is down (or the phone is having connectivity issues and points itself to B automatically). In this case when A goes down each phone will automatically failover to B when the next call is placed. There is still an issue of inbound calls, but most carriers will provide a mechanism to fail calls over as well (if the server is truly down). The regisration retry interval (to the best of my knowledge) is how long the phone will wait before attempting to re-register with servers of a higher priority after failing over. We tend to set this as a pretty low number because we want things to get back to the primary as soon as possible (since our carriers will fail back immediately for any new call coming in once the server is back up). -Steve ___ 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 phone failover using DNS SRV?
Rana Dutt wrote: Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP >adresses of A and B are associated with the same name, and both servers have equal priority and equal weight. In order to make calls through B after A goes down, do you have to wait as long as the registration retry > interval? Or can you make calls through B as soon as you pick up the phone and dial, because the INVITE >message through A fails, and the phone re-sends the INVITE through B? Thanks for any help. We've been trying this with Aastra 480i phones and SJ Phone without much luck so far. I do this with SIPura. I tell the box to use DNS SRV records. The first record points to the internal IP of my asterisk server (for use when I am at the home/office). The second one points to the external IP of my router that has port forwarding. I can roam pretty much anywhere without any reconfiguration. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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 PHONE
Try www.SIPphone.com or www.terracall.com Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ellafi FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] SIP PHONE Hi All, I just got a SIP phone and I would like to know where I could find service? Please helpThank you very much for your help Yahoo! SportsRekindle the Rivalries. Sign up for Fantasy Football NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 phone w/ XML browser
Brian Roy wrote: On 7/9/05, Dan Perik <[EMAIL PROTECTED]> wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian I should have tried it on my 501 before I went and opened my mouth. Sure enough, either it doesn't work, or I'm doing something wrong. The "Services" button is there, and the docs don't say anything about it not working, but even with it configured, it doesn't do anything. Seems to a be a "dead" button. Perhaps some firmware upgrade down the road will "turn it on". Looking through the archives I saw someone report that it did work on the 600, though. - Dan ___ 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 phone w/ XML browser
I use the polycom sip 500's with *. They are great. It also has a services buttons for XML services. I haven't looked in to using it just yet. Peace out, BrianOn 7/9/05, Mike Clark <[EMAIL PROTECTED]> wrote: Pavel Jezek wrote:> thank you Brian,> but seems, that Polycom phones are not very good option for general> corporate use and even not for use with asterisk (* explicitly> unsupported!), look: >> from voipsupply.com:> Please Note: Polycom phones are not supported under Asterisk Open> Source PBX.>> from Polycom FAQ:> Can the phones be used independent of a Technology Partner's platform? > No. In order to support full business phone features, the SoundPoint> IP is required to operate in conjunction with Partners' IP PBX...> Can the phones support LDAP directories?> Currently there is no support for directories like LDAP. > Is there a web browser built into the phone?> Polycom does not currently support this capability.>> I found avaya phones, that have nice features as I mentioned before> (e.g. xml browser) , any experiance with avaya SIP phones and their cost? > PJ>Well, we have over 100 Polycom phones deployed with Asterisk in acorporate environment and they are working extremely well.___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Brian McManus --(801) 652-5667 ___ 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 phone w/ XML browser
Pavel Jezek wrote: thank you Brian, but seems, that Polycom phones are not very good option for general corporate use and even not for use with asterisk (* explicitly unsupported!), look: from voipsupply.com: Please Note: Polycom phones are not supported under Asterisk Open Source PBX. from Polycom FAQ: Can the phones be used independent of a Technology Partner’s platform? No. In order to support full business phone features, the SoundPoint IP is required to operate in conjunction with Partners’ IP PBX... Can the phones support LDAP directories? Currently there is no support for directories like LDAP. Is there a web browser built into the phone? Polycom does not currently support this capability. I found avaya phones, that have nice features as I mentioned before (e.g. xml browser) , any experiance with avaya SIP phones and their cost? PJ Well, we have over 100 Polycom phones deployed with Asterisk in a corporate environment and they are working extremely well. ___ 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 phone w/ XML browser
thank you Brian, but seems, that Polycom phones are not very good option for general corporate use and even not for use with asterisk (* explicitly unsupported!), look: from voipsupply.com: Please Note: Polycom phones are not supported under Asterisk Open Source PBX. from Polycom FAQ: Can the phones be used independent of a Technology Partner’s platform? No. In order to support full business phone features, the SoundPoint IP is required to operate in conjunction with Partners’ IP PBX... Can the phones support LDAP directories? Currently there is no support for directories like LDAP. Is there a web browser built into the phone? Polycom does not currently support this capability. I found avaya phones, that have nice features as I mentioned before (e.g. xml browser) , any experiance with avaya SIP phones and their cost? PJ Brian Roy wrote: On 7/9/05, Dan Perik <[EMAIL PROTECTED]> wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian ___ 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 phone w/ XML browser
On 7/9/05, Dan Perik <[EMAIL PROTECTED]> wrote: > PJ, > > You should check out the Polycom 500/501/600. I'm quite sure it has all > that (although I don't use all of what you listed). > IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian ___ 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 phone w/ XML browser
PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). - Dan Pavel Jezek wrote: > Still looking for cheaper (under $250,-) alternative to cisco 7940 > with features needed for corporate use, mainly: > - shared phone book (e.g. via LDAP or XML browser in phone) > - in-line power > - missed/dialed/received numbers > - integrated switch (voice VLAN support) > > I found only aastara/sayson phone (and Intracom/Netphone in the past), > that has xml services anounced, but still not available, so any other > recommendation? Seems, that xml minibrowser isn't obvious even in high > end phone, but I think that via this function can be phone very > extensible... > thanks > PJ > ___ 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 Phone Config Generator
We use a mix of Cisco/Polycom phones. I just keep generic template files and make a copy of them for each new phone. It would be fairly trivial to put a webserver on the same server with the tftpd. In php or perl it would be fairly trivial to make a webpage that copied template files, replaced default values, and wrote out SIPmac.cnf or mac.cfg and mac-ext.cfg. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Max Clark Sent: Tuesday, June 28, 2005 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Phone Config Generator Hi all, Cisco/Polycom phones will pull their configuration via a tftp server to help manage mass deployments of phone systems. Are there any config generators available that will create the file for the tftp server? TIA, Max ___ 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 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 Phone Config Generator
I believe that [EMAIL PROTECTED] will create cisco files. - Original Message - From: "Max Clark" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, June 28, 2005 4:32 PM Subject: [Asterisk-Users] SIP Phone Config Generator > Hi all, > > Cisco/Polycom phones will pull their configuration via a tftp server to > help manage mass deployments of phone systems. Are there any config > generators available that will create the file for the tftp server? > > TIA, > Max > ___ > 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 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 Phone Recommendations?
Just want to let everyone know that even if there changing it out to the new 501 it's still on of the best. Remember that people are still buying the Cisco 7960G which is being phased out as well. The IP-500 works and works very well. I know that there price will be going down soon once there are some supplies of the IP-501. But if you need a phone now it is a very good one for the price. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, May 19, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Recommendations? On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote: >Ariel, > > It's probably not a good idea to reccomend the IP 500/300 anymore. >They are being phased out by Polycom because they (and the IP 300) only >have 2mb of flash, and Polycom is looking to standardize on 4mb for >their firmware (which the IP 600 has had since day one). > > If you are going to buy a Polycom now, get an IP 600, or, wait for the >301's or 501's. Don't say I didn't warn you! Good advice!. BTW, I LOVE my IP600's. I also kinda like the Zultys 4x4/4x5.The hardware and software is good but their support arrangement is terrible. They provide no end user support at all. Period. They rely upon their dealers to provide all support, but then they're ok with signing up dealers that know nothing about the products. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list 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 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 Phone Recommendations?
Kristian Kielhofner wrote: > It's probably not a good idea to reccomend the IP 500/300 anymore. > They are being phased out by Polycom because they (and the IP 300) > only have 2mb of flash, and Polycom is looking to standardize on 4mb > for their firmware (which the IP 600 has had since day one). > > If you are going to buy a Polycom now, get an IP 600, or, wait for > the 301's or 501's. Don't say I didn't warn you! It looks like the only features the 300 and 500 currently don't support, but the 301, 501, and 600 *do* support, is HTTPS/FTPS encrypted provisioning. Am I wrong about this? Obviously this may change in the future, but the 300 and 500 haven't suddenly become any *less* capable than they were last week. They may just not get *more* capable in the future. ___ 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 Phone Recommendations?
On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote: >Ariel, > > It's probably not a good idea to reccomend the IP 500/300 anymore. >They are being phased out by Polycom because they (and the IP 300) only >have 2mb of flash, and Polycom is looking to standardize on 4mb for >their firmware (which the IP 600 has had since day one). > > If you are going to buy a Polycom now, get an IP 600, or, wait for the >301's or 501's. Don't say I didn't warn you! Good advice!. BTW, I LOVE my IP600's. I also kinda like the Zultys 4x4/4x5.The hardware and software is good but their support arrangement is terrible. They provide no end user support at all. Period. They rely upon their dealers to provide all support, but then they're ok with signing up dealers that know nothing about the products. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list 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 Phone Recommendations?
Snom make good gear. Not cheap though. PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Thursday, 19 May 2005 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Recommendations? On Wed, 18 May 2005, John Mensel wrote: > Hi all. I'm in the process of putting together a new Asterisk system > as a proof-of-concept, and wanted to see which SIP phones all of you > had the best luck using with Asterisk. I've just come off a very > trying experience with some Cisco 7960s, and am looking for something > else to round out the phones on our network. Try the Grandstream GXP-2000. With the upcoming firmware it fits our needs except for the receptionist. Note that we use headsets instead of speakerphones except in conference rooms. If a good two-way speakerphone is needed you should look at other phones. The price is hard to beat. Peter ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 Phone Recommendations?
On Wed, 18 May 2005, John Mensel wrote: > Hi all. I'm in the process of putting together a new Asterisk system as a > proof-of-concept, and wanted to see which SIP phones all of you had the best > luck using with Asterisk. I've just come off a very trying experience with > some Cisco 7960s, and am looking for something else to round out the phones > on our network. Try the Grandstream GXP-2000. With the upcoming firmware it fits our needs except for the receptionist. Note that we use headsets instead of speakerphones except in conference rooms. If a good two-way speakerphone is needed you should look at other phones. The price is hard to beat. Peter ___ 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 Phone Recommendations?
Ariel Batista wrote: The receptionist phone is going to be a hard one. We use Flash Operator Panel. Works great. Now about the phones for all around great phone we are using the Polycom IP-500 which is in my view one of the top of the line phones. For el cheapo well we are using one that is yes cheap but also pretty good. Sipura 841 is filling this bill for us. As well as for some of the users where we want something a little better then the Sipura we use the Polycom IP-300. Ariel Ariel, It's probably not a good idea to reccomend the IP 500/300 anymore. They are being phased out by Polycom because they (and the IP 300) only have 2mb of flash, and Polycom is looking to standardize on 4mb for their firmware (which the IP 600 has had since day one). If you are going to buy a Polycom now, get an IP 600, or, wait for the 301's or 501's. Don't say I didn't warn you! -- Kristian Kielhofner ___ 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 Phone Recommendations?
The receptionist phone is going to be a hard one. We use Flash Operator Panel. Works great. Now about the phones for all around great phone we are using the Polycom IP-500 which is in my view one of the top of the line phones. For el cheapo well we are using one that is yes cheap but also pretty good. Sipura 841 is filling this bill for us. As well as for some of the users where we want something a little better then the Sipura we use the Polycom IP-300. Ariel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Wednesday, May 18, 2005 10:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP Phone Recommendations? Hi all. I'm in the process of putting together a new Asterisk system as a proof-of-concept, and wanted to see which SIP phones all of you had the best luck using with Asterisk. I've just come off a very trying experience with some Cisco 7960s, and am looking for something else to round out the phones on our network. This is a small setup, for no more than 20 users total. We need at least one of them to be a "receptionist" phone, the sort that calls can be routed from throughout the network, and then several more garden-variety handsets for regular users. A couple of el-cheapos to stick in out-of -the-way, little used spots would also be nice, if there are any good ones out there. Any phones that you've loved? Any that you've hated? Your input will be most welcome. Cheers, John Mensel ___ 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 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 Phone Recommendations?
This may not be what you are looking for, but I have had pretty good success with the X-Lite phone. I am not sure if you are looking for software SIP phones. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Mensel Sent: Wednesday, May 18, 2005 7:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP Phone Recommendations? Hi all. I'm in the process of putting together a new Asterisk system as a proof-of-concept, and wanted to see which SIP phones all of you had the best luck using with Asterisk. I've just come off a very trying experience with some Cisco 7960s, and am looking for something else to round out the phones on our network. This is a small setup, for no more than 20 users total. We need at least one of them to be a "receptionist" phone, the sort that calls can be routed from throughout the network, and then several more garden-variety handsets for regular users. A couple of el-cheapos to stick in out-of -the-way, little used spots would also be nice, if there are any good ones out there. Any phones that you've loved? Any that you've hated? Your input will be most welcome. Cheers, John Mensel ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.12 - Release Date: 5/17/2005 ___ 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 Phone Compatability
On Wed, 20 Apr 2005, Daniel Dziubanski wrote: > Greg, > > Are you using AMP? No. > And If so, you have any tips and tricks on how to easily manage phones via a > amp "plugin/fix"? No. The Polycom phones will provision themselves via FTP using XML files. It probably wouldn't be hard to write. In fact, SipX is supposed to have a decent Polycom provisioning engine that could be used as a basis. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 Phone Compatability
Greg, Are you using AMP? And If so, you have any tips and tricks on how to easily manage phones via a amp "plugin/fix"? - Original Message - From: "Greg Boehnlein" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 20, 2005 2:37 PM Subject: Re: [Asterisk-Users] SIP Phone Compatability On Wed, 20 Apr 2005, Daniel Salama wrote: Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. I've had very good luck with the Polycom SoundPoint IP 500 phones. They are rock solid, have great speakerphones, all of the features you list and more... Work just great with Asterisk. My clients love them, and at $185 / phone it is hard to beat. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.1 - Release Date: 4/20/2005 ___ 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 Phone Compatability
The Zultys Zip phone is crap though. As with the 841, no PoE No speakerphone No display I am unable to get the message waiting indication to work I am unable to get it to register with Asterisk, though I can place and receive calls There is no wall mounting bracket, and support doesn't have a clue what that even might be. Perhaps the higher priced ones are better. Frankly, for the under 100 dollar market, the Budgetone is the best value. The speakerphone works, they have a backlit display, and the only serious shortcoming is that the web configuration screen can't be accessed even to view settings when the phone is in use. John Novack Kerry Garrison wrote: The 841 is lacking in programmable buttons, it is an entry level phone. All additional features have to be accessed via access codes. For example, to transfer a call, dial #, voicemail dial *98, etc. The Zultys phones have programmable buttons for those features. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Compatability The SPA-841 doesn't seem to have conference call feature. This is extremely important. - Daniel On Apr 20, 2005, at 11:12 AM, Kerry Garrison wrote: I currently use an SPA-841 on my desk and don't have any problems with it http://www.geekgazette.com/index.php?option=com_content&task=view&id=2 4 I have been looking at these phones and they have more "office features" http://www.zultystechnologies.com/index.jsp? tab=product_list&type=phones -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Mason (Lists) Sent: Wednesday, April 20, 2005 3:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Compatability The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 12:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Phone Compatability Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. Any suggestions? Thanks, Daniel ___ 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 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 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 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 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 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 Phone Compatability
On Wed, 20 Apr 2005, Daniel Salama wrote: > Every once in a while I read messages about people having problems with > certain models of SIP phones, some of them being well known models. > > I'm interested in purchasing new SIP phones for my office and wanted to > know which brand/model is most stable with Asterisk, which allows most > office features. These features should include: multiple-line > appearances (at least 3), call conference, blind and non-blind > transfer, memory buttons or speed dials, voice message light indicator, > speaker phone, mute, redial, caller-id display. Anything on top of > these features is a plus but not really a requirement. I've had very good luck with the Polycom SoundPoint IP 500 phones. They are rock solid, have great speakerphones, all of the features you list and more... Work just great with Asterisk. My clients love them, and at $185 / phone it is hard to beat. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 Phone Compatability
The 841 is lacking in programmable buttons, it is an entry level phone. All additional features have to be accessed via access codes. For example, to transfer a call, dial #, voicemail dial *98, etc. The Zultys phones have programmable buttons for those features. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Compatability The SPA-841 doesn't seem to have conference call feature. This is extremely important. - Daniel On Apr 20, 2005, at 11:12 AM, Kerry Garrison wrote: > I currently use an SPA-841 on my desk and don't have any problems with > it > http://www.geekgazette.com/index.php?option=com_content&task=view&id=2 > 4 > > I have been looking at these phones and they have more "office > features" > http://www.zultystechnologies.com/index.jsp? > tab=product_list&type=phones > > -Kerry > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Chris > Mason > (Lists) > Sent: Wednesday, April 20, 2005 3:36 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] SIP Phone Compatability > > The Sipura SPA-841 has everything except memory buttons but has a > directory and speeddials so I don't think that's so important. Cheap > and well made, although if the speaker phone is very important, get > Polycoms, it's the business they are best in. > > Chris Mason > www.anguillaguide.com > > >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Daniel >> Salama >> Sent: Wednesday, April 20, 2005 12:14 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [Asterisk-Users] SIP Phone Compatability >> >> Every once in a while I read messages about people having problems >> with certain models of SIP phones, some of them being well known >> models. >> >> I'm interested in purchasing new SIP phones for my office and wanted >> to know which brand/model is most stable with Asterisk, which allows >> most office features. These features should include: multiple-line >> appearances (at least 3), call conference, blind and non-blind >> transfer, memory buttons or speed dials, voice message light >> indicator, speaker phone, mute, redial, caller-id display. Anything >> on top of these features is a plus but not really a requirement. >> >> Any suggestions? >> >> Thanks, >> Daniel >> >> ___ >> 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 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 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 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 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 Phone Compatability
The SPA-841 doesn't seem to have conference call feature. This is extremely important. - Daniel On Apr 20, 2005, at 11:12 AM, Kerry Garrison wrote: I currently use an SPA-841 on my desk and don't have any problems with it http://www.geekgazette.com/index.php?option=com_content&task=view&id=24 I have been looking at these phones and they have more "office features" http://www.zultystechnologies.com/index.jsp? tab=product_list&type=phones -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, April 20, 2005 3:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Compatability The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Wednesday, April 20, 2005 12:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Phone Compatability Every once in a while I read messages about people having problems with certain models of SIP phones, some of them being well known models. I'm interested in purchasing new SIP phones for my office and wanted to know which brand/model is most stable with Asterisk, which allows most office features. These features should include: multiple-line appearances (at least 3), call conference, blind and non-blind transfer, memory buttons or speed dials, voice message light indicator, speaker phone, mute, redial, caller-id display. Anything on top of these features is a plus but not really a requirement. Any suggestions? Thanks, Daniel ___ 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 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 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 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 Phone Compatability
I currently use an SPA-841 on my desk and don't have any problems with it http://www.geekgazette.com/index.php?option=com_content&task=view&id=24 I have been looking at these phones and they have more "office features" http://www.zultystechnologies.com/index.jsp?tab=product_list&type=phones -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, April 20, 2005 3:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Compatability The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Daniel > Salama > Sent: Wednesday, April 20, 2005 12:14 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] SIP Phone Compatability > > Every once in a while I read messages about people having problems > with certain models of SIP phones, some of them being well known > models. > > I'm interested in purchasing new SIP phones for my office and wanted > to know which brand/model is most stable with Asterisk, which allows > most office features. These features should include: multiple-line > appearances (at least 3), call conference, blind and non-blind > transfer, memory buttons or speed dials, voice message light > indicator, speaker phone, mute, redial, caller-id display. Anything on > top of these features is a plus but not really a requirement. > > Any suggestions? > > Thanks, > Daniel > > ___ > 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 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 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 Phone Compatability
The Sipura SPA-841 has everything except memory buttons but has a directory and speeddials so I don't think that's so important. Cheap and well made, although if the speaker phone is very important, get Polycoms, it's the business they are best in. Chris Mason www.anguillaguide.com > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Daniel Salama > Sent: Wednesday, April 20, 2005 12:14 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] SIP Phone Compatability > > Every once in a while I read messages about people having > problems with certain models of SIP phones, some of them > being well known models. > > I'm interested in purchasing new SIP phones for my office and > wanted to know which brand/model is most stable with > Asterisk, which allows most office features. These features > should include: multiple-line appearances (at least 3), call > conference, blind and non-blind transfer, memory buttons or > speed dials, voice message light indicator, speaker phone, > mute, redial, caller-id display. Anything on top of these > features is a plus but not really a requirement. > > Any suggestions? > > Thanks, > Daniel > > ___ > 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 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 phone extensions at a remote site
If you're using SIP I think what you want is canreinvite=yes which means the two remote user clients can talk directly to each other. Asterisk disappears from the loop which means no accounting. I think NAT causes problems in this scenario also. More details on the wiki Regards Cameron - Original Message - From: "cmould" <[EMAIL PROTECTED]> To: Sent: Saturday, April 09, 2005 7:48 PM Subject: [Asterisk-Users] sip phone extensions at a remote site I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. ___ 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 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 Phone binary
X-lite for Linux. http://www.xten.com/apps/xprolinuxbeta/ --- Klaus Peras <[EMAIL PROTECTED]> wrote: > Hey there, > > does anybody know a SIP-Client that I only have to > unpack and can run it > on Linux just like SJPhone, except SJPhone?? > > I need a Softphone for a Levigo Thin-Client, wich is > not having a compiler. > > > regards > > Klaus Peras > > > > > > begin:vcard > fn:Klaus Peras > n:Peras;Klaus > org:HOB;Netzwerk Support > adr;quoted-printable:;;Schwaderm=C3=BChlstrasse > 3;Cadolzburg;Bayern;90556;Germany > email;internet:[EMAIL PROTECTED] > tel;work:09103 / 715 - 329 > url:http://www.hob.de > version:2.1 > end:vcard > > > ___ > 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] SIP Phone with headset
The Uniden UIP200 is a decent phone with a headphone jack if the Sipura doesn't appeal to you. -Original Message- From: Thibault Lamy [mailto:[EMAIL PROTECTED] Sent: Thursday, February 24, 2005 11:00 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Phone with headset Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault ___ 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 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 Phone with headset
On Thu, February 24, 2005 11:59 am, Thibault Lamy said: > Do anyone have any experience with SIP phone that support > a headset ? We have Budgetone phones but we need headsets. Just deployed a batch of Sipura SPA-841's. Headset jack is standard so a few of us are using our cell-phone headsets and love it. Headset / handset / speakerphone all work well. Wish display could tilt to make it more readable. Not a big fan of the soft rubber buttons but I learned after a couple mistakes and it works fine. Got them new for under $100 each making the display and buttons non-issues IMHO. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ 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 Phone with headset
Grandstream are supposed to be releasing a BT103 ? Its a 100 series phone with headphone jack... when, I couldn't say though. Thibault Lamy wrote: Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault ___ 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 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 Phone with headset
Thibault Lamy wrote: Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. I am using Snom 190's with Snom head sets and like them a lot. On my list of things to do is using the Snom with a Labtec PC headset and see if that works as well (would be cheaper). -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ 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 Phone
The free phones I have heard of are soft phones... X-Lite is excellent... Cheap phones to test with, Grandstream is cheap but like the man said, you get what you pay for. eBay is a good source for cheap phones to test with but cheap is relative I consider cheap as sub $100. You can pick up a couple of models of Cisco and Polycom for 100-150. W -Original Message- From: Huddleston, Robert [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 8:04 AM To: 'Shaun Ewing'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone Anyone know where we could get a cheap sip phone... We've been playing with an Innomedia MGCP and SIP adapters and failing - so thinking that testing with a real phone might be good.. Robert A. Huddleston, KF4BYY IT Support Analyst Cavalier Telephone LLC. (Cell) 804.400.3686 [EMAIL PROTECTED] -Original Message- From: Shaun Ewing [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 11:04 AM To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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] SIP Phone
Anyone know where we could get a cheap sip phone... We've been playing with an Innomedia MGCP and SIP adapters and failing - so thinking that testing with a real phone might be good.. Robert A. Huddleston, KF4BYY IT Support Analyst Cavalier Telephone LLC. (Cell) 804.400.3686 [EMAIL PROTECTED] -Original Message- From: Shaun Ewing [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 11:04 AM To: Michael Bielicki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 ___ 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
Do you have a price range? 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. 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. Cheers, Wiley -Original Message- From: Michael Bielicki [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Cisco 7940 :) - Original Message - From: Phil Siegrist <[EMAIL PROTECTED]> Date: Wed, 22 Sep 2004 10:15:57 -0400 Subject: [Asterisk-Users] SIP Phone To: [EMAIL PROTECTED] Hi All, I am look for recommendations for a good SIP phone, specifically with a good speaker phone. I have tried the SNOM 100 and the speaker phone quality is quite poor. Can any one share there experiences with this. Much Appreciated, Phil ___ 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 -- Michael Bielicki ___ 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 The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer ___ 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
Cisco 7940 :) - Original Message - From: Phil Siegrist <[EMAIL PROTECTED]> Date: Wed, 22 Sep 2004 10:15:57 -0400 Subject: [Asterisk-Users] SIP Phone To: [EMAIL PROTECTED] Hi All, I am look for recommendations for a good SIP phone, specifically with a good speaker phone. I have tried the SNOM 100 and the speaker phone quality is quite poor. Can any one share there experiences with this. Much Appreciated, Phil ___ 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 -- Michael Bielicki ___ 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 -> PBX Phone
Seems to be alot of these questions on the mailing list recently. AUSTEL is the old name for the ACA, A-tick is the correct term for certification. It's only illegal if you connect to a carrier network without A-tick (you can get consent from them to connect without A-tick). The ACA has plently of info, such as http://www.aca.gov.au/consumer_info/fact_sheets/consumer_fact_sheets/fsc69.htm -Adam P J wrote: Thanks Paul. I've been getting conflicting information about Austel permits.. Can any one confirm that the card connecting Asterix to an existing PABX does not require Austel approval? Therefore, I could use a simple 1 port Compatible X100P FXO Card (that doesn't have Austel approval)? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 3:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone The TDM400 is used for both PSTN and PABX -> PABX connections, from memory. The card only requires an Austel permit if it is to be connected to an outside line, from memory. Cost wise, you can get the TDM400 with 1 line for less than $200, or about $500 with 4 lines hooked up to it. Later, Paul Hales IT Support Adairs -Original Message- From: Phil Stevens [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Hi Paul, I have yet to find out the make and model of the PABX, I was just doing some general background research at this point. Is the same device (TDM400?) used for connecting both to PSTN and to other PABX's? I don't need to connect the Asterisk box to PSTN at this point, just to another PABX. I do know that the current PABX already has a tie-line that connects to a second PABX. If Asterisk connects to another PABX via FXO ports, does the card have to have an Austel permit? I do not want to connect Asterisk to PSTN, only to another PABX. Do I have to purchase an $800 4-port FXO Austel-certified card? Or, is there a cheaper option (1 port? for testing purposes) that will satisfy Austel requirements? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 2:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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 -> PBX Phone
Thanks Paul. I've been getting conflicting information about Austel permits.. Can any one confirm that the card connecting Asterix to an existing PABX does not require Austel approval? Therefore, I could use a simple 1 port Compatible X100P FXO Card (that doesn't have Austel approval)? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 3:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone The TDM400 is used for both PSTN and PABX -> PABX connections, from memory. The card only requires an Austel permit if it is to be connected to an outside line, from memory. Cost wise, you can get the TDM400 with 1 line for less than $200, or about $500 with 4 lines hooked up to it. Later, Paul Hales IT Support Adairs -Original Message- From: Phil Stevens [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Hi Paul, I have yet to find out the make and model of the PABX, I was just doing some general background research at this point. Is the same device (TDM400?) used for connecting both to PSTN and to other PABX's? I don't need to connect the Asterisk box to PSTN at this point, just to another PABX. I do know that the current PABX already has a tie-line that connects to a second PABX. If Asterisk connects to another PABX via FXO ports, does the card have to have an Austel permit? I do not want to connect Asterisk to PSTN, only to another PABX. Do I have to purchase an $800 4-port FXO Austel-certified card? Or, is there a cheaper option (1 port? for testing purposes) that will satisfy Austel requirements? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 2:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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 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] SIP Phone -> PBX Phone
The TDM400 is used for both PSTN and PABX -> PABX connections, from memory. The card only requires an Austel permit if it is to be connected to an outside line, from memory. Cost wise, you can get the TDM400 with 1 line for less than $200, or about $500 with 4 lines hooked up to it. Later, Paul Hales IT Support Adairs -Original Message- From: Phil Stevens [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Hi Paul, I have yet to find out the make and model of the PABX, I was just doing some general background research at this point. Is the same device (TDM400?) used for connecting both to PSTN and to other PABX's? I don't need to connect the Asterisk box to PSTN at this point, just to another PABX. I do know that the current PABX already has a tie-line that connects to a second PABX. If Asterisk connects to another PABX via FXO ports, does the card have to have an Austel permit? I do not want to connect Asterisk to PSTN, only to another PABX. Do I have to purchase an $800 4-port FXO Austel-certified card? Or, is there a cheaper option (1 port? for testing purposes) that will satisfy Austel requirements? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 2:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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 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 -> PBX Phone
Hi Paul, I have yet to find out the make and model of the PABX, I was just doing some general background research at this point. Is the same device (TDM400?) used for connecting both to PSTN and to other PABX's? I don't need to connect the Asterisk box to PSTN at this point, just to another PABX. I do know that the current PABX already has a tie-line that connects to a second PABX. If Asterisk connects to another PABX via FXO ports, does the card have to have an Austel permit? I do not want to connect Asterisk to PSTN, only to another PABX. Do I have to purchase an $800 4-port FXO Austel-certified card? Or, is there a cheaper option (1 port? for testing purposes) that will satisfy Austel requirements? Thanks. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Friday, September 17, 2004 2:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP Phone -> PBX Phone Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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
RE: [Asterisk-Users] SIP Phone -> PBX Phone
Good to see another Australian user on the list! You could set up a card with some FXO ports (TDM400?) and use those lines to hook up the Asterisk box to your existing PABX. But I am sure someone else will come up with a _much_ more clever solution. Later, PaulH Melbourne -Original Message- From: P J [mailto:[EMAIL PROTECTED] Sent: Friday, 17 September 2004 12:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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] SIP Phone -> PBX Phone
What type of existing PABX do you have (Make and Model) What interfaces can you use to connect to your PABX, ie analog tie lines, E1/ISDN, anything else? Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of P J Sent: Friday, 17 September 2004 12:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone -> PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ 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] SIP Phone recommendation for Receptionist
Try the netweb-301 Hard IP Phone attached as a Pic Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Thompson Sent: Sunday, August 22, 2004 11:54 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Phone recommendation for Receptionist el Flynn wrote: > Hi there, > > I've got an installation where there's 12 POTS line incoming into *, > and am trying to get some insight as to which VoIP hard phone would > be most suitable for this scenario. > > Other than the incoming lines, the receptionist would need the normal > "keyphone" type stuff -- call pickup, park, hold, forward etc. > > What would you guys recommend? How about a touch screen LCD display running the Asterisk Flash Operator Panel? Or mabe a Tablet PC running Asterisk Flash Operator Panel? Jim James H. Thompson [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 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 recommendation for Receptionist
The expansion module is NOT supported with SIP. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jeremy Bogan > Sent: Sunday, August 22, 2004 7:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP Phone recommendation for > Receptionist > > > I've got an installation where there's 12 POTS line > incoming into *, > > and am trying to get some insight as to which VoIP hard > phone would be > > most suitable for this scenario. > > What would you guys recommend? > > A Cisco 7960 with the 7914 expansion module [ > http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/ind > ex.html ] > > -- > jeremy bogan[ [EMAIL PROTECTED] ] > segment publishing - design.develop.host > > ___ > 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] SIP Phone recommendation for Receptionist
el Flynn wrote: > Hi there, > > I've got an installation where there's 12 POTS line incoming into *, > and am trying to get some insight as to which VoIP hard phone would > be most suitable for this scenario. > > Other than the incoming lines, the receptionist would need the normal > "keyphone" type stuff -- call pickup, park, hold, forward etc. > > What would you guys recommend? How about a touch screen LCD display running the Asterisk Flash Operator Panel? Or mabe a Tablet PC running Asterisk Flash Operator Panel? Jim James H. Thompson [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
Re: [Asterisk-Users] SIP Phone recommendation for Receptionist
Jeremy Bogan wrote: I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/index.html ] The last I knew the 7914 only worked in SCCP not SIP. Jeremy McNamara ___ 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 recommendation for Receptionist
I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/index.html ] -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ 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 configuration problem
What phone do you have? On Fri, 16 Jul 2004 11:59:39 +0500, atif <[EMAIL PROTECTED]> wrote: > I am configuring a sip-phone, receing calls, excellent voice quality. but it does > not place calls, please, can some one sort out. > > here is my debug output, and below that is sip-debug, > > Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 > Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' > fesponse 1: Found > Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 > Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'UCWmUU1tF0s6roEx' of > Response 2: Found > Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'iiasPlzFribMJMcW' > Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'UCWmUU1tF0s6roEx' > > **SIP-DEBUG** > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS > Max-Forwards: 70 > From: chinee ;tag=Zlq179E4Jf8KX2lB > To: 13 > Call-ID: 1e020TNnX5IvcvFu > CSeq: 1 INVITE > Contact: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 221 > > v=0 > o=- 0 0 IN IP4 192.168.0.187 > s=- > c=IN IP4 192.168.0.187 > t=0 0 > m=audio 1400 RTP/AVP 0 8 4 18 0 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:0 telephone-event > > 12 headers, 11 lines > Using latest request as basis request > Sending to 192.168.0.187 : 5060 (non-NAT) > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 4 > Found RTP audio format 18 > Found RTP audio format 0 > Peer RTP is at port 192.168.0.187:0 > Found description format PCMU > Found description format PCMA > Found description format G723 > Found description format G729 > Found description format telephone-event > Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - > audio=0x109(G723|ALAW|G729A)/video=0x0(EMPTY), combined - 0x109(G723|ALAW|G729A) > Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) > Reliably Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS > From: chinee ;tag=Zlq179E4Jf8KX2lB > To: 13 ;tag=as51de164a > Call-ID: 1e020TNnX5IvcvFu > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="50b81cdd" > Content-Length: 0 > > to 192.168.0.187:5060 > Scheduling destruction of call '1e020TNnX5IvcvFu' in 15000 ms > Found user 'chinee' > > Atif > > > Sent via the WebMail system at convergence.com.pk > > ___ > 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] sip phone problem
--- Antonio Diego <[EMAIL PROTECTED]>
escribió: > >Hi,
> >First you need to upgrade to the latest CVS and
> then
> insert a second /
> >third priority line with hangup in the dialplan.
> >Regds
> >Vivian Alan
>
> Hi Alan, thanks for your help.
>
> My hardware is:
> -5 budgetone 100
> -2 handytone-286
> -Asterisk server running on Pentium IV, RAM 1GB,
> RedHat 8.0.
>
>
> I've just upgraded Asterisk and modified
> extensions.conf. My extensions.conf:
>
> ; extensions.conf
> [globals]
> ;static=yes
> EXTEN106=SIP/sip1
> EXTEN107=SIP/sip2
> EXTEN108=SIP/sip3
> EXTEN109=SIP/sip4
> EXTEN110=SIP/sip5
> EXTEN111=SIP/sip6
> EXTEN112=SIP/sip7
>
> [telefonos]
> include => todos
>
> [todos]
> exten => _1XX,1,NoOp()
> exten => _1XX,2,Dial(${EXTEN${EXTEN}})
> exten => _1XX,3,Hangup
> exten => _1XX,4,Hangup
> exten => _1XX,103,Hangup
> exten => h,1,Hangup
> exten => t,1,Hangup
>
> My sip.conf
>
> [general]
> disallow=all
> allow=alaw
> ;tos=lowdelay
> bindaddr=0.0.0.0
> nat=no
> language=es
>
> [sip1]
> type=friend
> secret=sip1
> host=dynamic
> defaultip=172.16.190.100
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip1" <106>
>
> [sip2]
> type=friend
> secret=sip2
> host=dynamic
> defaultip=172.16.190.101
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip2" <107>
>
> [sip3]
> type=friend
> secret=sip3
> host=dynamic
> defaultip=172.16.190.102
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip3" <108>
>
> [sip4]
> type=friend
> secret=sip4
> host=dynamic
> defaultip=172.16.190.103
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip4" <109>
>
> [sip5]
> type=friend
> secret=sip5
> host=dynamic
> defaultip=172.16.190.104
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip5" <110>
>
> [sip6]
> type=friend
> secret=sip6
> host=dynamic
> defaultip=172.16.190.105
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip6" <111>
>
> [sip7]
> type=friend
> secret=sip7
> host=dynamic
> defaultip=172.16.190.106
> dtmfmode=rfc2833
> context=telefonos
> callerid="sip7
>
> And the problem is still the same: Asterisk doesn't
> detect the budgetone hangup.
>
> The configuration of the Grandstream phones are:
> -for sip1
> http://tonidiego.webcindario.com/sip1.htm
> -for sip2
> http://tonidiego.webcindario.com/sip2.htm
> -for sip3
> http://tonidiego.webcindario.com/sip3.htm
> -for sip4
> http://tonidiego.webcindario.com/sip4.htm
> -for sip5
> http://tonidiego.webcindario.com/sip5.htm
> -for sip6
> http://tonidiego.webcindario.com/sip6.htm
> -for sip7
> http://tonidiego.webcindario.com/sip7.htm
>
>
> Thanks in advance
>
>
OK. I solved the problem.
MUCHAS GRACIAS SERGIO.
We added the global parameter "port=5060" in sip.conf.
That's all.
Everything is working OK.
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RE: [Asterisk-Users] sip phone problem
>Hi,
>First you need to upgrade to the latest CVS and then
insert a second /
>third priority line with hangup in the dialplan.
>Regds
>Vivian Alan
Hi Alan, thanks for your help.
My hardware is:
-5 budgetone 100
-2 handytone-286
-Asterisk server running on Pentium IV, RAM 1GB,
RedHat 8.0.
I've just upgraded Asterisk and modified
extensions.conf. My extensions.conf:
; extensions.conf
[globals]
;static=yes
EXTEN106=SIP/sip1
EXTEN107=SIP/sip2
EXTEN108=SIP/sip3
EXTEN109=SIP/sip4
EXTEN110=SIP/sip5
EXTEN111=SIP/sip6
EXTEN112=SIP/sip7
[telefonos]
include => todos
[todos]
exten => _1XX,1,NoOp()
exten => _1XX,2,Dial(${EXTEN${EXTEN}})
exten => _1XX,3,Hangup
exten => _1XX,4,Hangup
exten => _1XX,103,Hangup
exten => h,1,Hangup
exten => t,1,Hangup
My sip.conf
[general]
disallow=all
allow=alaw
;tos=lowdelay
bindaddr=0.0.0.0
nat=no
language=es
[sip1]
type=friend
secret=sip1
host=dynamic
defaultip=172.16.190.100
dtmfmode=rfc2833
context=telefonos
callerid="sip1" <106>
[sip2]
type=friend
secret=sip2
host=dynamic
defaultip=172.16.190.101
dtmfmode=rfc2833
context=telefonos
callerid="sip2" <107>
[sip3]
type=friend
secret=sip3
host=dynamic
defaultip=172.16.190.102
dtmfmode=rfc2833
context=telefonos
callerid="sip3" <108>
[sip4]
type=friend
secret=sip4
host=dynamic
defaultip=172.16.190.103
dtmfmode=rfc2833
context=telefonos
callerid="sip4" <109>
[sip5]
type=friend
secret=sip5
host=dynamic
defaultip=172.16.190.104
dtmfmode=rfc2833
context=telefonos
callerid="sip5" <110>
[sip6]
type=friend
secret=sip6
host=dynamic
defaultip=172.16.190.105
dtmfmode=rfc2833
context=telefonos
callerid="sip6" <111>
[sip7]
type=friend
secret=sip7
host=dynamic
defaultip=172.16.190.106
dtmfmode=rfc2833
context=telefonos
callerid="sip7
And the problem is still the same: Asterisk doesn't
detect the budgetone hangup.
The configuration of the Grandstream phones are:
-for sip1
http://tonidiego.webcindario.com/sip1.htm
-for sip2
http://tonidiego.webcindario.com/sip2.htm
-for sip3
http://tonidiego.webcindario.com/sip3.htm
-for sip4
http://tonidiego.webcindario.com/sip4.htm
-for sip5
http://tonidiego.webcindario.com/sip5.htm
-for sip6
http://tonidiego.webcindario.com/sip6.htm
-for sip7
http://tonidiego.webcindario.com/sip7.htm
Thanks in advance
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RE: [Asterisk-Users] sip phone problem
Hi, First you need to upgrade to the latest CVS and then insert a second / third priority line with hangup in the dialplan. Regds Vivian Alan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 25, 2004 7:48 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #3893 - 15 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: ZAPTEL not loading on FC2 (Fran Boon) 2. Re: Meetme Options (new one) (Fran Boon) 3. Sip/IAX Clients for Linux ([EMAIL PROTECTED]) 4. Re: 11 instead of Star (Peter Corlett) 5. sip phone problem (=?iso-8859-1?q?Antonio=20Diego?=) 6. Re: Sip/IAX Clients for Linux ([EMAIL PROTECTED]) 7. RE: Sip/IAX Clients for Linux (Karl Dyson) 8. Troubles with Kphone (enano) 9. RE: 11 instead of Star (Paul Crick) 10. RE: 11 instead of Star (Paul Crick) 11. TerraCall Setting ([EMAIL PROTECTED]) 12. Sound card problem ([EMAIL PROTECTED]) 13. Answer App hanging in I4L (S.Murali Krishna) 14. [Fwd: Answer App hanging in I4L] (Murali Krishnan) 15. Re: Troubles with Kphone] (Murali Krishnan) --__--__-- Message: 1 Date: Tue, 25 May 2004 09:52:48 +0100 From: Fran Boon <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZAPTEL not loading on FC2 Reply-To: [EMAIL PROTECTED] Jorge Verastegui wrote: > I have successfully compiled the last cvs zaptel drives in FC2 box and > then load wcfxs module, but Kernel Freezes with zttool http://bugs.digium.com/bug_view_page.php?bug_id=0001704 F --__--__-- Message: 2 Date: Tue, 25 May 2004 09:56:47 +0100 From: Fran Boon <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Meetme Options (new one) Reply-To: [EMAIL PROTECTED] Ben Merrills wrote: > Seems like it would be a simple modification? > Where would I post a feature request like this? :-) bugs.digium.com Ensure summary starts with [request] F --__--__-- Message: 3 Date: Tue, 25 May 2004 11:07:12 +0200 To: [EMAIL PROTECTED] From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sip/IAX Clients for Linux Reply-To: [EMAIL PROTECTED] Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark --__--__-- Message: 4 Date: Tue, 25 May 2004 10:09:19 +0100 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 11 instead of Star From: Peter Corlett <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] On Mon, May 24, 2004 at 07:58:26PM -0700, Paul Crick wrote: [...] > *sighs* Yeah, that won't work.. which is a shame.. this goes back to the > whole debate of "should CLASS service codes be implemented in the dial > plan or the channel driver?" The most compelling reason to me to have them in the dial plan is that CLASS codes aren't universal. AFAICS, they're mainly limited to the USA and telcos that source cheap switches from the USA and don't bother to customise them. For my Asterisk setup, I'd rather my phone uses the same star codes as BT and GSM than some foreign standard that nobody here knows. > From memory and reading the mailing list for a while now, I think Mark's > dead against having these features in the dial plan, but I can't remember > why. Let me guess, is he American? ;) [...] > I think the way it was going to go was a flag which would allow you to > disable all channel driver features like this and rely on the dial plan to > implement the features. This is very much my preferred solution. If there is still some bizarre obligation to support alien phone standards in the channel drivers, we should have the option of disabling this undesired behaviour. --__--__-- Message: 5 Date: Tue, 25 May 2004 04:21:53 -0500 (CDT) From: =?iso-8859-1?q?Antonio=20Diego?= <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip phone problem Reply-To: [EMAIL PROTECTED] Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem connecting budgetones to Asterisk? Please help me. Thanks in advance. _ Do You Yahoo!? Inf
Re: [Asterisk-Users] SIP phone registering problem
First pass through the trace indicates all udp packets originating from 194.200.209.137 have incorrect checksums. However, the asterisk machine acknowledged the initial register packet with a 100 Trying, therefore it must be ignoring udp checksums. (Still curious why incorrect checksums are generated consistently.) It would appear that asterisk accepted the Register (with the 200 OK), but then .137 attempts another Register four seconds later. Might try type=user (instead of type=friend) and host= to bypass the register function and see if that works. > On Wednesday 07 April 2004 09:24, Richard Airlie wrote: > > On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: > > > download ethereal and take a peek at the packets on the wire. Without > > > something like that, no one is really going to be able to help you. > > > > I thought I'd chime in here with a packet dump from ethereal, since I'm also > using KPhone 4.02 and I feel there's maybe a bug in its registration code, > since I remember using KPhone 3.xx and having no problems. > > However, I would appreciate it if someone more experienced could cast their > eyes over this to make sure I'm not doing something stupid :) > > My Asterisk sip.conf is simply: > > [general] > port = 5060 > bindaddr = 0.0.0.0 > context = default > > [myphone] > type=friend > host=dynamic > context=default > qualify=1000 > > And the KPhone configuration is set as follows: > > User part of SIP URL: myphone > Host part of SIP URL: gdh > Outbound Proxy: tel > > 'gdh' is the name of my local PC and resolves to 194.200.209.137 > 'tel' is the remote Asterisk server at 213.2.4.46. > > I have made sure there is no firewall issue between the two by two catch-all > iptables -A FORWARD -s 194.200.209.137 -d 213.2.4.46 -j ACCEPT and > iptables -A FORWARD -d 194.200.209.137 -s 213.2.4.46 -j ACCEPT > at the firewall on each side. > > I've attached a gzipped snippet from ethereal in libpcap format of the failing > registration. > > If the list doesn't permit attachments, it's also available at > http://bum.net/sip.cap > > Cheers, > Gavin. > ---End of Original Message- ___ 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 registering problem
On Wednesday 07 April 2004 09:24, Richard Airlie wrote: > On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: > > download ethereal and take a peek at the packets on the wire. Without > > something like that, no one is really going to be able to help you. > I thought I'd chime in here with a packet dump from ethereal, since I'm also using KPhone 4.02 and I feel there's maybe a bug in its registration code, since I remember using KPhone 3.xx and having no problems. However, I would appreciate it if someone more experienced could cast their eyes over this to make sure I'm not doing something stupid :) My Asterisk sip.conf is simply: [general] port = 5060 bindaddr = 0.0.0.0 context = default [myphone] type=friend host=dynamic context=default qualify=1000 And the KPhone configuration is set as follows: User part of SIP URL: myphone Host part of SIP URL: gdh Outbound Proxy: tel 'gdh' is the name of my local PC and resolves to 194.200.209.137 'tel' is the remote Asterisk server at 213.2.4.46. I have made sure there is no firewall issue between the two by two catch-all iptables -A FORWARD -s 194.200.209.137 -d 213.2.4.46 -j ACCEPT and iptables -A FORWARD -d 194.200.209.137 -s 213.2.4.46 -j ACCEPT at the firewall on each side. I've attached a gzipped snippet from ethereal in libpcap format of the failing registration. If the list doesn't permit attachments, it's also available at http://bum.net/sip.cap Cheers, Gavin. sip.cap.gz Description: GNU Zip compressed data
Re: [Asterisk-Users] SIP phone registering problem
> On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: > > download ethereal and take a peek at the packets on the wire. Without > > something like that, no one is really going to be able to help you. > > Do you mean then that my SIP trace displayed at kphone looks otherwise OK -- > that the REGISTER request that kphone's sending out looks alright? > > Is there a single good resource describing the SIP protocol specification > so I know what I should be looking for? One good reference is: http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming_reference_guide_book0 9186a0080080221.html Since you indicated asterisk was not showing any activity from that register, it is highly likely the register activity isn't going to where you think it should, or something like that. I'd suspect that you really don't need the sip protocol reference noted above, just a clue from the packet trace what is really happening on the wire. SIP packets (on the wire) contain a substantial amount of plain text, so looking at two or three packets to see what is really happening doesn't require very much technical understanding or references. Rich ___ 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 registering problem
On Tue, Apr 06, 2004 at 05:52:51PM -0600, Rich Adamson wrote: > download ethereal and take a peek at the packets on the wire. Without > something like that, no one is really going to be able to help you. Do you mean then that my SIP trace displayed at kphone looks otherwise OK -- that the REGISTER request that kphone's sending out looks alright? Is there a single good resource describing the SIP protocol specification so I know what I should be looking for? thanks, Richard. ___ 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 registering problem
download ethereal and take a peek at the packets on the wire. Without something like that, no one is really going to be able to help you. > I am clearly doing something ridiculously wrong. > > Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are > unable to register. They keep trying and then time out. > > With the sip debug on in Asterisk nothing is logged. > Here is the trace from one of the phones (kphone): > > (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) > > sipclient: sending: 21:47:45.454 > > REGISTER sip:192.168.100.3 SIP/2.0 > Via: SIP/2.0/UDP 192.168.100.13;rport > CSeq: 4399 REGISTER > To: "sjphone2" > Expires: 900 > From: "sjphone2" > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" ;methods="INVITE, > MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" > > SipClient: Sending to '192.168.100.3:5060' > SipClient: Receiving message... > > SipClient: Received: 21:47:45.471 > - > REGISTER sip:192.168.100.3 SIP/2.0 > Via: SIP/2.0/UDP 192.168.100.13;rport > CSeq: 4399 REGISTER > To: "sjphone2" > Expires: 900 > From: "sjphone2" > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" ;methods="INVITE, > MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" > > SipCall: Incoming request > SipCall: New transaction created > SipTransaction: Incoming Request > SipTransaction: Retransmit 1 (4000) > > SipClient: Sending: 21:47:49.456 > > REGISTER sip:192.168.100.3 SIP/2.0 > Via: SIP/2.0/UDP 192.168.100.13;rport > CSeq: 4399 REGISTER > To: "sjphone2" > Expires: 900 > From: "sjphone2" > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" ;methods="INVITE, > MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" > > SipClient: Receiving message... > > SipClient: Received: 21:47:49.466 > - > REGISTER sip:192.168.100.3 SIP/2.0 > Via: SIP/2.0/UDP 192.168.100.13;rport > CSeq: 4399 REGISTER > To: "sjphone2" > Expires: 900 > From: "sjphone2" > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" ;methods="INVITE, > MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" > > > SipCall: Incoming request > SipTransaction: Incoming Request Retransmission > SipTransaction: Response Retransmission > SipTransaction: Retransmit 2 (4000) > > (and so it continues) > > Seems like the REGISTER messages are being recieved at Asterisk but > then just echoed back to the SIP phone? What am I doing wrong? > > thanks! > > Richard. > ___ > 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 ---End of Original Message- ___ 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 with push display?
At 7:54 AM -0600 3/31/04, Rich Adamson wrote: From: Rich Adamson <[EMAIL PROTECTED]> To: Asterisk-a-users-list <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Sip phone with push display? Anyone know of a business class sip hard phone that includes a quality display capable of supporting "push" data (maybe Polycom?). Something like... VM: 3 msgs OurStock (1:43pm): 59.5 somewhere on the display that can be updated (pushed) from a server? Rich The Polycom IP600 (and IP500?) support it, though I have not worked directly with those units and the subscribe/notify syntax yet. The Snom phone (220?) support it, and even have a nice operator console, so I'm very interested to see those ship Real Soon Now. The Sayson (Aastra?) IP phones support it, but they're also vaporware until next month. I saw the InterTel 8690 at VON, which I'm _certain_ supports it, but I think $1200 is a bit out of your budget range (though it is a VERY nice looking phone, even though it runs CE.) InterTel's other phones support it - models 8662 and 8620 probably will do what you want, but only with LEDs to say who's on the phone (no "big display" on those units.) The Cisco 79xx phones don't support "push", which really is disappointing. It would be trivial to do (though it _must_ be authenticated!) and would be a big asset if they had a NOTIFY method that allowed an XML URL to be sent to the phone. I've beaten my head on the table enough about Cisco's crippled SIP features; I'm shortly going to start to experiment with and possibly move to other platforms if they prove stable enough. I'm a big fan of Cisco equipment, but they haven't kept up with other vendors in the SIP software department, and they've never kept up with "transparency" from a purchasing perspective. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
Sure, here's my extension for paging on the intercom:
[ext-intercom-one];
exten => _87XXX,1,SetVar(ALERT_INFO="Ring Answer")
exten => _87XXX,2,Dial(SIP/${EXTEN:1},20,r)
exten => _87XXX,103,Congestion
exten => _87XXX,104,Congestion
exten => t,1,Hangup
Internally, I use a four digit extension here, starting with 7. When I
preface it with 8, it calls this extension, which sets the ALERT_INFO
variable and makes the phone work its auto-answer magic. If I dial my
partner with 7002, for example, it'll ring normally, go to voicemail,
etc. If I dial him with 87002, it beeps his office and his phone
automatically answers.
There's probably a better way to do this, but I've only had these phones
for a couple of weeks.
In answer to your other question about the bootrom, I think you need at
least the 2.4.0 bootrom to run the latest SIP software with these
features. I'm running 2.4.1. The SIP software version you must have to
do this is 1.1.0.
If you go to polycom's website and download the manual for these phones
and the 1.1.0 release notes, you can find out how to do all sorts of
tricks by manipulating the cfg files. (Sadly, you can only get the
manuals from Polycom. You have to get the bootrom and software from
your vendor)
John Baker
On Wed, 2003-12-31 at 08:24, mattf wrote:
> I've added it as a separate page:
> http://www.voip-info.org/wiki-Polycom+auto-answer+config linked from the
> Polycom phones page.
>
> Could you possibly send me a quick line or two(example code) on setting the
> ALERT_INFO variable in Asterisk?
>
> Thanks,
>
> MATT---
>
>
>
> -Original Message-
> From: mattf [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 31, 2003 7:57 AM
> To: '[EMAIL PROTECTED]'
> Subject: RE: [Asterisk-Users] SIP phone as intercom
>
>
> Cool, haven't looked that in depth into the new firmware(is that the 2.4.1
> firmware?) I'll have to try that.
> I'll post your instructions on the Wiki page later today.
>
> Thanks,
>
> MATT---
>
> -----Original Message-
> From: John Baker [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, December 31, 2003 3:07 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] SIP phone as intercom
>
>
> Hello, all
>
> Sorry to correct you on this Matt, but I am currently doing this with
> the Polycom 600 phones. You need the latest version of both the SIP
> software and bootrom to do it, (and that stuff ain't easy to get) but it
> is workable.
>
> The latest version of software provides for distinctive ring tones just
> like the Cisco 7960's have. It also provides for auto answer. It's
> kind of tricky to do, but you can make your phones auto answer by
> setting the Alert-Info variable in asterisk and messing with the xml
> configuration files, sip.cfg and ipmid.cfg.
>
> In the sip.cfg file, look for the line with these variables:
>
> voIpProt.SIP.alertInfo.1.class="8"...>
>
> In this real-world example, whenever I set ALERT_INFO to "Sales" in
> Asterisk, the Polycom matches on that word and calls up class 8 in
> ipmid.cfg.
>
> In ipmid.cfg, my class 8 line looks like this:
>
> se.rt.8.callWait="6" se.rt.8.mod="0">
>
> se.rt.8.type="ring" tells the Polycom phone which type of ring to use -
> which in this case is a regular ring and se.rt.8.ringer="11" tells the
> phone to ring with ringtone 11 with is the Triplet.
>
> I use this one for signaling a new incoming sales call to one of my
> three sales guys. The secretary transfers it to the sales department
> and their lines ring with the Triplet. I feel like Pavlov whenever I
> hear it.
>
> The other ring types are visual, answer and ring-answer. The one you
> want is ring-answer.
>
> Here's how I do it: Again in sip.cfg (actually part of the same line
> listed above)
>
> ...voIpProt.SIP.alertinfo.2.value="Ring Answer"
> voIpProt.SIP.alertInfo.2.class="4"...>
>
> and in ipmid.cfg (I just modified one of the existing ones to give me a
> High Double Trill ringtone)
>
> se.rt.4.timeout="1000"... se.rt.4.ringer="7"...>
>
> The se.rt.4.timeout="1000" tells the Polycom to ring for 1000
> milliseconds (one second) and then answer.
>
> I call it in Asterisk by setting the ALERT_INFO variable to "Ring
> Answer" whenever anybody pushes 8 plus the extension. It rings in to
> the extension and voila, I'm on speaker!
>
> By the way, for all you BOFH out there, you could actually use this
> feature as a somewhat surreptitious eavesdropping device by using a
> silent ring and a
Re: [Asterisk-Users] SIP phone as intercom
Wow! Thanks John for the detailed information. This is such an awesome system... and great support here, too. On Dec 31, 2003, at 12:07 AM, John Baker wrote: Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: In this real-world example, whenever I set ALERT_INFO to "Sales" in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: se.rt.8.type="ring" tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer="11" tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"...> and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) The se.rt.4.timeout="1000" tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to "Ring Answer" whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
I've added it as a separate page: http://www.voip-info.org/wiki-Polycom+auto-answer+config linked from the Polycom phones page. Could you possibly send me a quick line or two(example code) on setting the ALERT_INFO variable in Asterisk? Thanks, MATT--- -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 7:57 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] SIP phone as intercom Cool, haven't looked that in depth into the new firmware(is that the 2.4.1 firmware?) I'll have to try that. I'll post your instructions on the Wiki page later today. Thanks, MATT--- -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:07 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP phone as intercom Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: In this real-world example, whenever I set ALERT_INFO to "Sales" in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: se.rt.8.type="ring" tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer="11" tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"...> and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) The se.rt.4.timeout="1000" tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to "Ring Answer" whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: > Hello, > > It's all dependant upon the firmware of the phone(nothing to do with the PBX > or SIP currently). The documentation of the Polycom VOIP phones shows no way > of doing this currently but it is really just a matter of Polycom adding > this feature to their firmware in the future which we are pushing for. > People have gotten this to work with Cisco and Snom phones. > > MATT--- > > -Original Message- > From: Sean Adams [mailto:[EMAIL PROTECTED] > Sent: Tuesday, December 30, 2003 6:21 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] SIP phone as intercom > > > > (new asterisk user - currently setting up Polycom IP600 phones) > > Does anyone know if it's possible to make a sip phone instantly pick up > on speakerphone when a particular call comes in? Eg so that you can > quickly bother someone across the office without making them reach for > their phone? > > ___ > 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 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] SIP phone as intercom
Cool, haven't looked that in depth into the new firmware(is that the 2.4.1 firmware?) I'll have to try that. I'll post your instructions on the Wiki page later today. Thanks, MATT--- -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:07 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP phone as intercom Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: In this real-world example, whenever I set ALERT_INFO to "Sales" in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: se.rt.8.type="ring" tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer="11" tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"...> and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) The se.rt.4.timeout="1000" tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to "Ring Answer" whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: > Hello, > > It's all dependant upon the firmware of the phone(nothing to do with the PBX > or SIP currently). The documentation of the Polycom VOIP phones shows no way > of doing this currently but it is really just a matter of Polycom adding > this feature to their firmware in the future which we are pushing for. > People have gotten this to work with Cisco and Snom phones. > > MATT--- > > -Original Message- > From: Sean Adams [mailto:[EMAIL PROTECTED] > Sent: Tuesday, December 30, 2003 6:21 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] SIP phone as intercom > > > > (new asterisk user - currently setting up Polycom IP600 phones) > > Does anyone know if it's possible to make a sip phone instantly pick up > on speakerphone when a particular call comes in? Eg so that you can > quickly bother someone across the office without making them reach for > their phone? > > ___ > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: In this real-world example, whenever I set ALERT_INFO to "Sales" in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: se.rt.8.type="ring" tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer="11" tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"...> and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) The se.rt.4.timeout="1000" tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to "Ring Answer" whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: > Hello, > > It's all dependant upon the firmware of the phone(nothing to do with the PBX > or SIP currently). The documentation of the Polycom VOIP phones shows no way > of doing this currently but it is really just a matter of Polycom adding > this feature to their firmware in the future which we are pushing for. > People have gotten this to work with Cisco and Snom phones. > > MATT--- > > -Original Message- > From: Sean Adams [mailto:[EMAIL PROTECTED] > Sent: Tuesday, December 30, 2003 6:21 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] SIP phone as intercom > > > > (new asterisk user - currently setting up Polycom IP600 phones) > > Does anyone know if it's possible to make a sip phone instantly pick up > on speakerphone when a particular call comes in? Eg so that you can > quickly bother someone across the office without making them reach for > their phone? > > ___ > 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
RE: [Asterisk-Users] SIP phone as intercom
Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ 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] SIP Phone Tone
On Tue, 2003-10-14 at 20:14, Eric Wieling wrote: > Yes, of course. However, that would be a feature of the SIP phone, not > Asterisk, since Asterisk isn't providing the dialtone on your SIP phone, > the phone is doing that. Cisco 79XX can. You have to put "9," on the dialplan.xml -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone Tone
Yes, of course. However, that would be a feature of the SIP phone, not Asterisk, since Asterisk isn't providing the dialtone on your SIP phone, the phone is doing that. On Tue, 2003-10-14 at 16:28, Chris Hariga wrote: > Hi, > > si posible on SIP phones to have the dial tone after 9 like on the FXS card? > I set ignorepat => 9 on my extensions.conf... > > Best regards, > > Chris HARIGA > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
Peter Pauly wrote: > If Asterisk registers with a SIP long distance provider and > I make a call from an IP phone through Asterisk to that > LD provider, does the RTP (audio) traffic flow between the two > end points directly (normally the IP phone and the LD provider) or > does it flow through Asterisk? > > I'm asking because I have Asterisk running behind a NAT firewall > along with an IP Phone (software) and I'm trying to get it > working with Iconnecthere (ICH). I am able to register, connect > , but no audio. I have ports opened up on the firewall, but > they point to the Asterisk machine and not the IP phone machine. > In this scenario, any audio traffic would have to go through the > asterisk box to reach the IP phone. Is that how it works? SIP control connection usualy goes thru firewall. RTP - no. Just put the Asterisk on the machine with the firewall and it will work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
> I'm asking because I have Asterisk running behind a NAT firewall > along with an IP Phone (software) and I'm trying to get it > working with Iconnecthere (ICH). I am able to register, connect > , but no audio. I have ports opened up on the firewall, but > they point to the Asterisk machine and not the IP phone machine. > In this scenario, any audio traffic would have to go through the > asterisk box to reach the IP phone. Is that how it works? I was using a sniffer a few minutes ago to identify an issue between a cisco 7960 and ata186. The call setup occurs between the phones and asterisk on udp 5060 (both source and destination ports), but the actual conversation was directly between the phones (in at least this one example) on udp ports 23570 and 1, with 180 byte data payloads occuring approximately every 20 milliseconds. Another call between XLite and a Snom 200 used udp ports 8000 and 10018 directly between the phones. The above is only intended to point out the NATing issues associated with using voip through a firewall. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone to use with *
Sean, > Any recommendations on a hardware based SIP phone to use with *? > > I'm looking for something that would be common, as well as quick and easy > to source, somthing relatively quick and easy to configure. I'm very new to this as well, but with 20+ years of telephony and data network performance background. Between the Snom 200 and Cisco 7960 (both seem to work well), the Snom is much quicker to deploy, you don't need a tftp server (which is pretty much a requirement for the 7960), takes up less desktop space, etc, etc. Both phones provide roughly the same functionality, however if you have folks that like lots of buttons or like well-recognized names, the Cisco does a fine job. >From what I've seen on the list, there are many other choices as well, I just don't have any experience/knowledge of those. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone behind NAT
Hi Olaf, I've just started working on a SIP and RTP proxy to handle exactly this. I'm really just in proof of concept at the moment but just one hour ago I got a completely successful connection out over NAT in which both endpoints thought they were talking to the proxy. I should have the code posted in the next few days. So far it's only tested under Linux but it should work on Windows without too many problems. I'll post more info in the next few days but feel free to email me directly if you are inerested or haven't heard anything from me. Regards, Andrew Radke Olaf Menzel wrote: Hi all, I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP to the internal Snom100 IPadress and a port range forwarding of 16384 - 32768 (UDP) for the RTP traffic. Additionally I tried to change host = dynamic to host = myserver.dyndns.org to ensure the SIP traffic is going to my Linksys ADSL router and be forwarded to the internal SIP 100 phone. But all my effort did not have success. Any suggestions ?? regards Olaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
