Re: [asterisk-users] IEEE 802.1x capable sip phones
I called Cisco and they are so far the only vendor that offers it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Moskowitz Sent: Wednesday, January 09, 2008 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IEEE 802.1x capable sip phones Jeronimo Romero wrote: > > Does anyone know if sip phones from any of the major IP phone vendors > support 802.1x authentication? Any feedback would be greatly appreciated. > This is so unlikely. I worked on 802.1X and 802.11i. There is just too much overhead there. No way to meet the ITU 50ms disruption requirement. Plus it is a lot of code. Wait until 802.11r and/or 11s get done to get any real secure roaming. Rather implement SRTP. > > > > Thanks in advance. > > > > == > Jeronimo Romero > EUS Networks > Email: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > Cell: 917-332-7238 > Office: 212-624-5943 > Web: www.euscorp.com <http://www.euscorp.com> > == > > > > > > > > > > ___ > -- 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
[asterisk-users] IEEE 802.1x capable sip phones
Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] aastra phones with asterisk 1.2.17 - hangup after 20 seconds
Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message Im getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 783509378 (Critical Response) Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. The strange thing is that when I use an xten softphone this issue does not occur. Is this a SIP signaling issue? Any help would be appreciated. This issue does not occurr with any other ip phone on our network. ONLY THE AASTRAs. any ideas? ___ --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] Nice Transfer Feature
Just be careful with the sidecar. It was to be screwed on and the screws that come with the unit strip very easily. Make sure you have a nice electronics grade screwdriver with a long thin shaft or you'll have trouble with the side car. Another really nice feature of this phone is that the BLF for call parking and call park pickup work flawlessly with asterisk 1.4.1 and up. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Thursday, March 29, 2007 2:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Nice Transfer Feature I just noticed the Aastra 57i do something that I haven't seen before. I called from one phone (phone 1) to the 57i. I answered it. Then, I pressed Transfer and dialed the extension for the third phone (in this case a Cisco 7960 in Sip). I did not answer the Cisco, but noticed the caller ID was showing the Aastra (as expected). I hung up the Aastra to complete the transfer and noticed the caller ID on the Cisco was updated to show the Caller ID of the original caller (phone 1), just like a blind transfer would do. The point is that the caller ID was updating during the ringing. I had not seen a phone capable of this. This, along with the one touch parking and XML capabilities, is looking like the Aastra may be what I'm looking for in a receptionist phone. -- Lacy Moore Somewhere I wish I wasn't ___ --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] TE110P: Error ==> Asterisk died with code 1.
The revision on this card was new and my version of zaptel (1.2.11) did not support my pri signaling for that card. I upgraded zaptel through svn and all was good. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, February 28, 2007 12:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TE110P: Error ==> Asterisk died with code 1. On Wed, Feb 28, 2007 at 10:47:48AM -0500, Jeronimo Romero wrote: > Thank you all. Was a signaling issue. And for the benefit of those who will read the archive: how have you debugged it? how have you resolved it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] camp on off-line phone
It would be cool if you could add some kind of login script capability to nodes in sip.conf and iax.conf. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Sunday, March 18, 2007 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] camp on off-line phone Leif Neland wrote: > When phone A registers, I want phone B to ring, when picked up, it should > call phone A and connect the phones. > > Translated: When GF in Mexico powers up laptop where soft iax-phone > registers automatically, I want to talk to her asap :-) > > How to? I don't really know how to do this, but wouldn't it be easiest if she just called you as soon as she is online? Sorry for not being of any help. :-( Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] T1 cable for Digium T1/E1 Cards
So a regular cross over cable wouldn't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Sunday, March 18, 2007 10:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
Yes. At that point, you're looking for a T1 "cross-over".
The pinout is as follows:
1
4
RX/Ring/- <-->TX/Ring/-
2
5
RX/Tip/+ <-->TX/Tip/+
4
1
TX/Ring/- <-->RX/Ring/-
5
2
TX/Tip/+ <-->RX/Tip/+
3
3
Shield/Return/Ground
6
6
Shield/Return/Ground
On 3/18/07, Jeronimo Romero <[EMAIL PROTECTED]> wrote:
> I assume that I would need to cross these pins over if I were going
from
> t1 card to t1 card. Is this correct?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom
> Sent: Sunday, March 18, 2007 7:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards
>
> A common Cat5 straight through cable will work fine.
>
> T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for
> signals.
>
> A T1 loopback plug would be wired 1 to 4 and 2 to 5.
>
> They come in handy for testing T1 cards or for providing a hard loop
> for the telco.
>
> Tom
>
> At 05:42 PM 3/18/2007, you wrote:
> >Is there any technical difference between a T1 cable and a cat5e
patch
> >cable as far as using them with Digium T1/E1 cards? Can PRI circuits
> >terminating at a smart jack connect successfully to Digium cards
using
> >straight through CAT5e cables? If so, are they using all of the pins
in
> >the cable?
> >
> >Thanks in advance
> >___
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> >
> >asterisk-users mailing list
> >To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
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--
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http://www.btwtech.com/
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RE: [asterisk-users] T1 cable for Digium T1/E1 Cards
I assume that I would need to cross these pins over if I were going from t1 card to t1 card. Is this correct? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Sent: Sunday, March 18, 2007 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 cable for Digium T1/E1 Cards A common Cat5 straight through cable will work fine. T1s use 1 and 2 (rx tip and ring) and 4 and 5 (tx tip and ring) for signals. A T1 loopback plug would be wired 1 to 4 and 2 to 5. They come in handy for testing T1 cards or for providing a hard loop for the telco. Tom At 05:42 PM 3/18/2007, you wrote: >Is there any technical difference between a T1 cable and a cat5e patch >cable as far as using them with Digium T1/E1 cards? Can PRI circuits >terminating at a smart jack connect successfully to Digium cards using >straight through CAT5e cables? If so, are they using all of the pins in >the cable? > >Thanks in advance >___ >--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
[asterisk-users] T1 cable for Digium T1/E1 Cards
Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance ___ --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] TE110P: Error ==> Asterisk died with code 1.
Thank you all. Was a signaling issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, February 28, 2007 12:55 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TE110P: Error ==> Asterisk died with code 1. On Tue, Feb 27, 2007 at 08:01:41PM -0500, Jeronimo Romero wrote: > Running Asterisk 1.2.9. I just installed a TE110P card and configured > zaptel.conf & zapata.conf. The config files look right to me but I'm > getting the following error when trying to start asterisk: > > Asterisk died with code 1. > Automatically restarting Asterisk. > > Does anyone have any idea what is wrong with this configuration?? > Thanks in advance!!! What is the output of: cat /proc/zaptel/* -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE110P: Error ==> Asterisk died with code 1.
Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf [channels] context=from-pstn switchtype=national signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=yes group=0 signalling=pri_cpe context = from-pstn channel =>1-23 ====== Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == ___ --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] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
here's what I found on voip-info.org http://www.voip-info.org/wiki/index.php?page=Avaya+or+Lucent+Magix+Voice mail+Integration From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS??? FXS cards generate ring (you connect a "station" to it and it rings). FXO cards sink ring (they take ring from the office). If the Octel needs ring (which it most likely does), you would need an FXS card to generate ring for it to answer. An FXO would take ring from the vmail server, which, in context, doesn't make a lot of sense (vmail doesn't call the PBX, the PBX calls vmail). EKG From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Monday, February 05, 2007 8:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS??? Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct? ___ --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] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
Thanks. Is there a way I can log into the Merlin Magix to determine that? How else do I tell? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS??? FXS cards generate ring (you connect a "station" to it and it rings). FXO cards sink ring (they take ring from the office). If the Octel needs ring (which it most likely does), you would need an FXS card to generate ring for it to answer. An FXO would take ring from the vmail server, which, in context, doesn't make a lot of sense (vmail doesn't call the PBX, the PBX calls vmail). EKG From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Monday, February 05, 2007 8:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS??? Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct? ___ --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] asterisk server as a voicemail server forlegacy PBX -- FXO or FXS???
I believe it will be hooked up to extension lines. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Monday, February 05, 2007 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk server as a voicemail server forlegacy PBX -- FXO or FXS??? Will the Asterisk box be hooked up to external lines on the Merlin, or extension lines? External - FXS Extension - FXO later, PaulH On Mon, 2007-02-05 at 20:03 -0500, Jeronimo Romero wrote: > Hey All, > > > > I'll be configuring an asterisk box to be the voicemail server to an > old Merlin system which had an octel 100 voicemail server that is now > dying. > > My question is simple: do I need to stick an FXO card in the asterisk > box? My logic is that if the Merlin Magix system is actually > generating electrical current, then I would need to have an fxo card. > Is this correct? > > > > > ___ > --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
[asterisk-users] asterisk server as a voicemail server for legacy PBX -- FXO or FXS???
Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct? ___ --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] Dell Servers
We have the 2950. It came with only 2pcix ports. And if you need to power an fxs card, you need to route wires around. It wasn't easy to work with. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Thursday, February 01, 2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dell Servers Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse <[EMAIL PROTECTED]> wrote: > Hi, > > I was planning on getting a Dell PowerEdge 2950 for our new Asterisk > configuration. > But while searching for documentation about it and/or reported issues, I > found this: > > http://www.voip-info.org/wiki/view/Asterisk+hardware > WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, > which has been known to cause random locksup - if you plan on using a > Dell server, disable the onboard controller and purchase an addon > ethernet card. > > Does anyone has real experience ? > > Thanks, > > -- > Eric Rousse > System Administrator > 514.380.2992 > 450.655.1001 > 1.888.641.5800 > > Telmatik inc. > 204 Montarville, suite 250 > Boucherville, QC, Canada > J4B 6S2 > > www.telmatik.com > > > ___ > --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
[asterisk-users] caller id ring tones for Asterisk Phone
I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk PLAR
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown). The Oreilly (Asterisk: Future of Telephony) book mentions it in passing saying that all you need to enable it is to set immediate=yes in zapata.conf. Has anyone implemented this in brokerage trading environments? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cal recording with email
I'm trying to set on-demand call recording. Here's a snippet of the
pertinent dialplan. The purpose of this is to allow one user in
particular to be able to receive an email recording of the call
everytime he dials *91 + number. The problem is that the email is not
going out or being generated when I use the ${CALLFILENAME} variable.
When I use the actual file name of the gsm recording, the emails go out
without a problem.
[rec-tt-trunkdial]
exten=>_*91NXX.,1,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
exten=>_*91NXX.,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME
},m)
exten=>_*91NXX.,n,Set(CALLERID(num)=7188233325)
exten=>_*91NXX.,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN:2},,gtTr
)
exten=>_*91NXX.,n,Wait(5)
exten=>_*91NXX.,n,System(cat /etc/macro-text | mailx -a
/var/spool/asterisk/monitor/ ${CALLFILENAME}.gsm -s "Recorded"
[EMAIL PROTECTED])
exten=>_*91NXX.,n,Hangup()
This is my asterisk console output:
Connected to Asterisk 1.2.12.1 currently running on pbx (pid = 1999)
Verbosity is at least 3
-- Hungup 'IAX2/voicepulse02-8'
-- Executing Wait("SIP/1001-081d9b80", "2") in new stack
-- Executing System("SIP/1001-081d9b80", "cat /etc/macro-text |
mailx -a /var/spool/asterisk/monitor/20061208-103611:1001.gsm -s
"hello" [EMAIL PROTECTED]") in new stack
-- Executing Hangup("SIP/1001-081d9b80", "") in new stack
== Spawn extension (rec-tt-trunkdial, *912126245943, 7) exited
non-zero on 'SIP/1001-081d9b80'
Nothing actually happens. For testing I replaced the ${CALLFILENAME}
variable in the System() command with the actual recording name:
Like this in extensions.conf:
exten=>_*91NXX.,n,System(cat /etc/macro-text | mailx -a
/var/spool/asterisk/monitor/20061208-103611:1001.gsm -s "Recorded"
[EMAIL PROTECTED])
This worked fine so I'm guessing that there's something wrong I'm doing
when passing the ${CALLFILENAME} variable to the linux shell in
System().
Any help would be appreciated. Thanks in advance.
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[asterisk-users] iptables example
Hey everyone. I recenty installed a server at a datacenter offsite and the thing is getting hammered with invalid ssh logins so I decided to use some iptables. I included my ruleset here. I was wondering if I could get some feedback based on my ruleset from those of you using iptables in production systems. It seems to be working but some critique would be appreciated. Thanks #!/bin/sh # My system IP/set ip address of server SERVER_IP="x.x.x.x" # Flushing all rules iptables -F iptables -X # Setting default filter policy iptables -P INPUT DROP iptables -P OUTPUT DROP iptables -P FORWARD DROP # Allow unlimited traffic on loopback iptables -A INPUT -i lo -j ACCEPT iptables -A OUTPUT -o lo -j ACCEPT # Allow incoming ssh only from secure hosts iptables -A INPUT -p tcp -s x.x.x.x -d $SERVER_IP --sport 513:65535 --dport 22 -m state --state NEW,ESTABLISHED -j ACCEPT iptables -A INPUT -p tcp -s x.x.x.x -d $SERVER_IP --sport 513:65535 --dport 22 -m state --state NEW,ESTABLISHED -j ACCEPT #Allow http & Asterisk Related Traffic iptables -A INPUT -p tcp -i eth0 --dport 80 -m state --state NEW -j ACCEPT # SIP on UDP iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT # IAX2- iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX - iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT iptables -A INPUT -j DROP iptables -A OUTPUT -j ACCEPT ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] calls hang up even after Background() messageeventhough response timeout is set to 10 sec
The problem was that autofallthrough=yes was set in extensions.conf
> I'm experiencing a strange problem. My inbound calls are hanging up
> right after Background() message even though response timeout is set
to
> 10 sec.
>
> [voicepulseincoming]
>
> exten=>_X.,1,Answer
> exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
> exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1)
> exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)
>
> [after-business-hours]
>
> exten=>s,1,Answer
> exten=>s,n,Set(TIMEOUT(digit)=10)
> exten=>s,n,Set(TIMEOUT(response)=10)
> exten=>s,n,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
> exten=>s,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME},m)
> exten=>s,n,Background(outside-business-hours)
> exten=>s,n,Background(main-auto-attendant)
> exten=>i,1,Goto(after-business-hours,s,7)
> exten=> 411,1,Directory(default)
> exten=> a,1,Goto(after-business-hours,s,7)
> exten=> o,1,Goto(after-business-hours,s,7)
>
>
> The call hangs up without respecting the 10 second response timeout.
> I've seen people posting this issue but I haven't seen the solution.
> Any help would be greatly appreciated.
>
> The asterisk console spits out the following message:
>
>
> -- Playing 'outside-business-hours' (language 'en')
> -- Executing BackGround("IAX2/voicepulse01-1",
> "main-auto-attendant") in new stack
> -- Playing 'main-auto-attendant' (language 'en')
> == Auto fallthrough, channel 'IAX2/voicepulse01-1' status is
'UNKNOWN'
> -- Hungup 'IAX2/voicepulse01-1'
>
>
>
>
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"Those that sacrifice essential liberty to obtain a little temporary
safety
deserve neither liberty nor safety." -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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[asterisk-users] calls hang up even after Background() message eventhough response timeout is set to 10 sec
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set to
10 sec.
[voicepulseincoming]
exten=>_X.,1,Answer
exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1)
exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)
[after-business-hours]
exten=>s,1,Answer
exten=>s,n,Set(TIMEOUT(digit)=10)
exten=>s,n,Set(TIMEOUT(response)=10)
exten=>s,n,SetVar(CALLFILENAME=${TIMESTAMP}:${CALLERIDNUM})
exten=>s,n,Monitor(gsm,/var/spool/asterisk/monitor/${CALLFILENAME},m)
exten=>s,n,Background(outside-business-hours)
exten=>s,n,Background(main-auto-attendant)
exten=>i,1,Goto(after-business-hours,s,7)
exten=> 411,1,Directory(default)
exten=> a,1,Goto(after-business-hours,s,7)
exten=> o,1,Goto(after-business-hours,s,7)
The call hangs up without respecting the 10 second response timeout.
I've seen people posting this issue but I haven't seen the solution.
Any help would be greatly appreciated.
The asterisk console spits out the following message:
-- Playing 'outside-business-hours' (language 'en')
-- Executing BackGround("IAX2/voicepulse01-1",
"main-auto-attendant") in new stack
-- Playing 'main-auto-attendant' (language 'en')
== Auto fallthrough, channel 'IAX2/voicepulse01-1' status is 'UNKNOWN'
-- Hungup 'IAX2/voicepulse01-1'
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RE: [asterisk-users] Asterisk On FreeBSD
I've installed on 6.1 it from ports with ztdummy without an issue. I've never used zaptel hardware on it though. Had some issues with meetme and ztdummy but all worked out. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of J. Oquendo > Sent: Wednesday, November 22, 2006 1:43 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk On FreeBSD > > Dumpolid Exeplish wrote: > > Hi, > > Has anyone installed Asterisk on FreeBSD? i need help/steps on this task > > > > > > ___ > > --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 > > > > http://archives.free.net.ph/message/20060618.125548.f385ddf1.en.html > > Complete how to > > -- > > J. Oquendo > http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 > sil . infiltrated @ net http://www.infiltrated.net > > The happiness of society is the end of government. > John Adams ___ --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] Grandstream GXP2000 -- What's the Catch?
Thanks for the input. I take it the snoms support both BLF & intercom? ====== Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Wednesday, November 15, 2006 9:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch? They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. We only do the Snom 320 or 360's now and are just as easy to configure and have alot of great options as well. On 11/15/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --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 -- Tom Vile ___ --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] simple mainmenu ivr tones not recognized
How is your DTMF mode configured in sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Perkins Sent: Wednesday, November 15, 2006 7:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] simple mainmenu ivr tones not recognized I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the tones to be recognized during the background( ) the playback and background files play, but asterisk doesn't do anything when I start pushing keys - I've tried it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice] exten => s,1,Answer exten => s,2,Playback(pbx-candles-welcome) exten => s,3,Background(pbx-candles-mainmenu) exten => 1,1,dial,SIP/101|45|r exten => 1,2,voicemail, exten => 1,3,hangup() exten => 2,1,dial,SIP/101|45|r exten => 2,2,voicemail, exten => 2,3,hangup() exten => 3,1,dial,SIP/102|45|r exten => 3,2,voicemail, exten => 3,3,hangup() exten => 4,1,Goto(from-broadvoice,s,2) [from-broadvoice2] exten => s,1,Answer exten => s,2,Playback(pbx-candled-route,skip),2 exten => s,3,dial,SIP/101|45|r exten => s,4,voicemail,1112 exten => s,5,hangup() [internal] exten => 101,1,dial,SIP/101|45|r exten => 101,2,voicemail, exten => 102,1,dial,SIP/101|45|r exten => 102,2,voicemail,1112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intercom function on eyebeam xten softphones.
We are currently using xten eyebeam soft phones for our laptops at work. We would like to know if it would be possible to configure the phone to auto pickup when it reads the sip header: "Call-Info: answer-after=0" Is this possible with this soft-phone or any other soft-phone ? thanks in advance. JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP2000 -- What's the Catch?
We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --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] Dual Wan Router with Failover
We've had great results with Astrocom powerlink for load balancing outbound wan connections. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Dean Collins > Sent: Tuesday, November 14, 2006 12:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: [EMAIL PROTECTED] > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > Thanks for the reply matthew, basically I've been looking at going to a > dmz model for a while as currently everything runs through a single > sbs2003 server and when it's churning drives doing something sometimes > audio errors occur. > > > Cheers, > > Dean > > > > -Original Message- > > From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, 14 November 2006 11:41 AM > > To: Dean Collins > > Cc: Asterisk-Users > > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > > > There are several dual-WAN routers with load balancing and > failover, > > including the Xincom Twin-WAN series that I have tested OK with SIP > (as > > NAT): http://www.xincom.com/twinwan.php . Their other products > probably > > work, too. > > > > Keep in mind that load balancing on these devices assigns each > > TCP/IP > > *connection* to its own WAN interface. So a large transfer on a single > > connection is limited by the bandwidth of whichever interface it's > > started on, even if the transfer starts slow enough to get assigned to > a > > smaller bandwidth interface, then expands to require the bandwidth > from > > the other WAN. The tech to de/multiplex streams works "well" only when > > connecting to a single router endpoint, over relatively few hops that > > can lengthen unpredictably the path some deplexed tackets travel. UDP > > works better, but it still doesn't really work that well. What works > > well is assigning different WANs to different apps' traffic, using > > multiple WANs for failover, or just accepting that these techs are > > better than nothing, and offer cheap ways to at least avoid a single > > point of failure in the WAN scheme. > > > > > > On Tue, 2006-11-14 at 08:07 -0700, > > [EMAIL PROTECTED] wrote: > > > Date: Tue, 14 Nov 2006 10:07:41 -0500 > > > From: "Dean Collins" <[EMAIL PROTECTED]> > > > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > > > Message-ID: > > > > > <[EMAIL PROTECTED]> > > > Content-Type: text/plain; charset="US-ASCII" > > > > > > Hi Jason, > > > I was looking for an external solution outside of my asterisk box so > > > that I can load balance my other website/email traffic as well. > > > > > > > > > Cheers, > > > > > > Dean > > > > > > > > > > -Original Message- > > > > From: > > > [EMAIL PROTECTED] [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Jason > > > > Sent: Tuesday, 14 November 2006 11:00 AM > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > If you don't mind using linux, linux can do some fairly intense > load > > > > balancing all built in. Check out the Linux Virtual Server > project. > > > As > > > > for WAN failover, if you again don't mind using linux, you can > > > script > > > a > > > > simple ping to the internet (I would ping at least 3 hosts) and if > > > that > > > > fails, fail to your second ISP. You can also do some crazy fun > > > stuff > > > > with linux advance routing and bonding. > > > > > > > > Jason > > > > The place where you made your stand never mattered, > > > > only that you were there... and still on your feet > > > > > > > > > > > > > > > > Dean Collins wrote: > > > > > > > > > > Are you looking for load balancing or failover. > > > > > > > > > > > > > > > > > > > > Also is there a cheaper way of
RE: [asterisk-users] same extension on softphones and hardphones
Is this inherently an issue with sip? Is it possible for a sip server to
actually ring two different sip registration from the same account or is
this not possible under any sip enabled pbx?
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Sunday, November 12, 2006 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] same extension on softphones and
hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
> Sorry if you see this message repeated twice. I would like to set up
> hard phones and softphones with the same extension so that when
anybody
> in the company dials an extension, each user's hardphone and softphone
> will ring at the same time. I've tried setting this up before, but I
> noticed that the last sip device to register with the same extension
is
> the only one that rings when the extension is dialed. The sip devices
> they will be using are Grandstream GXP2000 desktop phones and Xten
> Eyebeam softphones. Each user will have one of each. What is the best
> way to accomplish this?
>
>
> Xten eyebeam ext 110 \
> \
>--> Asterisk 1.2.8
> /
> GXP2000 phone ext 110 /
One possible solution is to have one sip account for each _device_, not
extension; say "sip110h" and "sip110s" for the "110"-user.
Then use the "dial" command in your extensions.conf like
exten => _1XX,1,Dial(SIP/sip${EXTEN}s&SIP/sip${EXTEN}h)
This will cause parallel ringing phones. First come first serve.
Hth
Anselm
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[asterisk-users] same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the extension is dialed. The sip devices they will be using are Grandstream GXP2000 desktop phones and Xten Eyebeam softphones. Each user will have one of each. What is the best way to accomplish this? Xten eyebeam ext 110 \ \ --> Asterisk 1.2.8 / GXP2000 phone ext 110 / Need both phones to ring when extension 110 is dialed. Is this possible without creating ring groups? Thanks in advance. JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] announcing inbound PSTN calls
I’m running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere? What is the best approach to doing this? Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a snippet of my extensions.conf
[default]
exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten=>_11XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
include=>record
include=>parkedcalls
include=>voicepulseoutgoing
include=>conferences
include=>voicemail
[macro-stdexten]
exten=>s,1,Dial(${ARG2},20,t)
exten=>s,2,Goto(s-${DIALSTATUS},1)
exten=>s-NOANSWER,1,Voicemail(u${ARG1})
exten=>s-NOANSWER,2,Goto(default,s,1)
exten=>s-BUSY,1,Voicemail(b${ARG1})
exten=>s-BUSY,2,Goto(default,s,1)
exten=>_s-.,1,Goto(s-NOANSWER,1)
exten=>a,1,VoicemailMain(${ARG1})
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
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RE: [asterisk-users] blind transfers with IP Polycom 501
Thanks so mAre there any drawbacks to this? This is the digitmap I ended up using and it seems to work: [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[1-9]xxxT -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Sunday, October 29, 2006 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 Yes, digitmap... If you just want to allow any digit pattern, use this digitmap: xx.T x -> Any valid digit . -> 0 or more ocurences of previous charracter T -> Default timeout (3 seconds) Any digit followed by a 3 second timeout will match. You can include pattern to match * and #. xx.T|*x.T|#x.T Julian. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: > Do you mean the digitmap?? > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Sunday, October 29, 2006 5:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 > > For the soft buttons to work the way you want it, make sure you got > the Polycom dialplan setup right. > > On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: > > > > > > > > > > I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only > > problem I'm experiencing is the following: I can't seem to get blind > > transfers to work with my Polycom 501 phones Either through the feature > > code or the soft keys. > > > > > > > > > > > > Feature code blind transfers: > > > > I set up a feature map in features.conf like this: > > > > blindxfer => # > > > > This works for all my softphones, just not the 501 phones. > > > > > > > > Soft key Blind Transfers: > > > > Then I tried blind transfers through the phone like this: > > > > Transfer à Blind Key à Extension. > > > > Here's the problem. If I enter a 2 digit extension, it works. > > > > Example: Transfer à Blind Key à 70 (this parks my calls without a problem) > > > > If I try to blind transfer an extension with three or more digits, the phone > > cancels the blind transfer. > > > > Is there something obvious I'm missing here?? > > > > > > > > Thanks in advance. > > > > > > > > > > > > > > > > > > ___ > > --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 ___ --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] blind transfers with IP Polycom 501
It was the digit map. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Sunday, October 29, 2006 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] blind transfers with IP Polycom 501 Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: > > > > > I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only > problem I'm experiencing is the following: I can't seem to get blind > transfers to work with my Polycom 501 phones Either through the feature > code or the soft keys. > > > > > > Feature code blind transfers: > > I set up a feature map in features.conf like this: > > blindxfer => # > > This works for all my softphones, just not the 501 phones. > > > > Soft key Blind Transfers: > > Then I tried blind transfers through the phone like this: > > Transfer à Blind Key à Extension. > > Here's the problem. If I enter a 2 digit extension, it works. > > Example: Transfer à Blind Key à 70 (this parks my calls without a problem) > > If I try to blind transfer an extension with three or more digits, the phone > cancels the blind transfer. > > Is there something obvious I'm missing here?? > > > > Thanks in advance. > > > > > > > > > ___ > --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] blind transfers with IP Polycom 501
Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: > > > > > I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only > problem I'm experiencing is the following: I can't seem to get blind > transfers to work with my Polycom 501 phones Either through the feature > code or the soft keys. > > > > > > Feature code blind transfers: > > I set up a feature map in features.conf like this: > > blindxfer => # > > This works for all my softphones, just not the 501 phones. > > > > Soft key Blind Transfers: > > Then I tried blind transfers through the phone like this: > > Transfer à Blind Key à Extension. > > Here's the problem. If I enter a 2 digit extension, it works. > > Example: Transfer à Blind Key à 70 (this parks my calls without a problem) > > If I try to blind transfer an extension with three or more digits, the phone > cancels the blind transfer. > > Is there something obvious I'm missing here?? > > > > Thanks in advance. > > > > > > > > > ___ > --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
[asterisk-users] blind transfers with IP Polycom 501
I’m running Asterisk 1.2.8 with Polycom ip501’s xten softphones The only problem I’m experiencing is the following: I can’t seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here’s the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I’m missing here?? Thanks in advance. ___ --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] WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]
Has anyone tried RedFone?? It is supposed to offload a lot of that bus overhead to the external unit doing TDMoE. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John covici > Sent: Thursday, September 28, 2006 12:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] WAS: 64 analog phones NOW: Selection > criteria and recipie for a good Asterisk install [long] > > OK, pardon my ignorance -- but what can you tune on such a system? > How does Linux handle separate buses? > > Thanks. > > on Thursday 09/28/2006 Colin Anderson([EMAIL PROTECTED]) > wrote > > >I concur with your approach, but "Tier 1" means as little here as it > > >does when evaluating Internet backbone carriers. could you expand on > > >what evaluation criteria you use? I'm going to be pre-speccing some > > >stuff myself this month... > > > > Sorry I should have been more clear. A good Asterisk install needs a > > holistic approach to use a hippy dippy phrase. A Tier 1 server, which > is a > > midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, > IBM, > > am I missing someone?) is usually highly optimized for bus bandwidth > > although that design was intended for a different use - usually massive > disk > > I/O. As well, a Tier 1 server will have two seperate, independent PCI > buses > > and this to me is a critical feature - it allows you to completely > separate > > your TDM traffic from network, disk I/O etc. On my big production > Netfinity, > > I took great care to ensure the Digium cards were all on their lonesome > on a > > single bus, and everything else on the other bus. This is how I can run > two > > TE110's in a single box with no problems. zttest does not give me 100% > all > > the time, but on the other hand it *never* drops below 99.9987%, even > under > > load. I selected this Netfinity because of the obvious care put into > it's > > design, but the specs are unimpressive: quad Xeon 700's. CPU is over > rated > > for Asterisk, IMO unless you are doing tons of transcoding and if you > are > > doing that, then your design is flawed. > > > > Anyway, the holistic approach (to go on a small rant for the newbie > lurkers) > > be summed up as follows: > > > > 1. Good box, see above > > 2. Good LAN - this is so critical and so often overlooked in the day > and age > > of guys crimping their own cables and running $150 switches. You can't > do > > that, and if you do, you do so at your own peril. Managed swiches, > > professional cable installation. This is not a problem for me since I > *am* a > > professional cable installer but I have actually witnessed people > making > > patch cables with a flat blade screwdriver and a hammer! > > 3. Tuning of the LAN - VLAN's are good. QoS packets are good. Switches > that > > honor the QoS packets are good. > > 4. Handset selection - this is another biggie. I've selected Snom > 360's, and > > yes they have warts, but they are feature rich for the price and Snom > is > > really good about revising firmware. When you select handsets, GET YOUR > > USERS INVOLVED. > > 5. Tuning of Asterisk box itself - this cannot be under emphasized. > This is > > a very important step and tuning methodologies vary according to > distro, > > skill of the admin, and particular circumstances. I've learned *way* > more > > than I ever wanted to about processor affinity sinc I started using > > Asterisk. > > 6. Termination of PSTN. Basically I would never do an Asterisk install > where > > I was forced to do something stupid like aggregate a dozen Centrex > lines or > > some mickey mouse deal with FXO ATA's or whatever except for a hobby or > > prototype install. PRI, BRI, IAX or SIP, don't mess around with > anything > > else. > > 7. Relationship with provider. What is their SLA? Is it the incumbent > or the > > clec? An incumbent will be more expensive and more difficult to deal > with > > but they will tend to be more reliable. A clec will be cheaper and they > will > > be way more accomodating but you will most likely not get five 9's from > > them. A VoIP provider should never be trusted, period. You will not get > five > > nines
RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8
Yes. It this is the opensource poundkey from rpath. I just installed madplay instead of dealing with mpg123. Works like a charm. Is there any downside to madplay that that I should know about?? Here my musiconhold.conf file: > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of brandon kruz > Sent: Tuesday, September 19, 2006 10:46 PM > To: asterisk-users@lists.digium.com > Subject: RE: [asterisk-users] MOH distorted on Pound Key Linux on > asterisk1.2.8 > > is this open source poundkey? > > and can i see your moh conf?? > > im guessing its open source pk since you mentioned the asterisk 1.2.8 > part. > > and also is it the default MOH or your own cooked up version?? > also i recommend, if not necessary the EXACT mpg version described in the > conf > (321 0.9?) something similar, please try this, but first of all before we > go > that extreme. > > lets see your conf's and if you made your own music, or default MOH > >From: "Jeronimo Romero" <[EMAIL PROTECTED]> > >Reply-To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >To: > >Subject: [asterisk-users] MOH distorted on Pound Key Linux on asterisk > >1.2.8 > >Date: Tue, 19 Sep 2006 20:34:20 -0400 > >MIME-Version: 1.0 > >Received: from lists.digium.com ([69.16.138.164]) by > >bay0-mc5-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Tue, > >19 Sep 2006 17:36:28 -0700 > >Received: from digium-69-16-138-164.phx1.puregig.net (localhost > >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 81F371C787;Tue, > 19 > >Sep 2006 17:34:20 -0700 (MST) > >Received: from psmtp.com (exprod8mx39.postini.com [64.18.3.139])by > >lists.digium.com (Postfix) with SMTP id 379154005for > >;Tue, 19 Sep 2006 17:34:15 -0700 (MST) > >Received: from source ([216.254.77.227]) by > >exprod8mx39.postini.com([64.18.7.10]) with SMTP; Tue, 19 Sep 2006 > 17:34:25 > >PDT > >X-Message-Info: LsUYwwHHNt39D7KiqZIglXEhMN6zJlkqI6ZJRfxWfWQ= > >X-Original-To: asterisk-users@lists.digium.com > >Delivered-To: asterisk-users@lists.digium.com > >Content-class: urn:content-classes:message > >X-MimeOLE: Produced By Microsoft Exchange V6.5 > >X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: MOH distorted on > Pound > >Key Linux on asterisk 1.2.8 > >Thread-Index: AcbcTIOMWMuXx9+PRq6pnRtp3I737w== > >X-pstn-levels: (S:49.04751/99.9 FC:95.5390 LC:95.5390 R:95.9108 > >P:95.9108M:94.9308 C:98.6951 ) > >X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c > >X-pstn-addresses: from <[EMAIL PROTECTED]> [db-null] X-BeenThere: > >asterisk-users@lists.digium.com > >X-Mailman-Version: 2.1.5 > >Precedence: list > >List-Id: Asterisk Users Mailing List - Non-Commercial > >Discussion > >List-Unsubscribe: > ><http://lists.digium.com/mailman/listinfo/asterisk- > users>,<mailto:asterisk-users- > [EMAIL PROTECTED]> > >List-Archive: <http://lists.digium.com/pipermail/asterisk-users> > >List-Post: <mailto:asterisk-users@lists.digium.com> > >List-Help: <mailto:[EMAIL PROTECTED]> > >List-Subscribe: > ><http://lists.digium.com/mailman/listinfo/asterisk- > users>,<mailto:[EMAIL PROTECTED]> > >Errors-To: [EMAIL PROTECTED] > >Return-Path: [EMAIL PROTECTED] > >X-OriginalArrivalTime: 20 Sep 2006 00:36:29.0792 (UTC) > >FILETIME=[D05B7E00:01C6DC4C] > > > >Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium > >site. > > > >Uname output: > > > >Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005 > >i686 athlon i386 GNU/Linux > > > > > > > > > > > >It didn't come with mpg123 so I downloaded it from the internet. MOH > >works, but it is terribly loud and mistorted. Tried running under > >quitemp3 profile but it didn't help. > > > > > > > >I feel like there is something I may be missing here. Any ideas??? > > > >Thanks in advance. > > > > > > > >Jeronimo. > > > > > > > > > > > >== > > > >Jeronimo Romero > > > >EUS Networks > > > >Email: [EMAIL PROTECTED] > > > >Cell: 917-332-7238 > > > >Office: 212-624-5943 > > > >Web: www.euscorp.com > > > >== > > > > > > > > > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >aster
RE: [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8
This is pound key linux from rpath. I don't see a source directory. That is why I think I must be missing something. >>-Original Message- >>From: [EMAIL PROTECTED] [mailto:asterisk-users->>[EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling >>Sent: Tuesday, September 19, 2006 10:49 PM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [asterisk-users] MOH distorted on Pound Key Linux on >>asterisk1.2.8 >>Remove mpg123. In the Asterisk source directory type "make mpg123" I >>believe that "make install" is required to install it. >>Jeronimo Romero wrote: > Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium > site. > > Uname output: > > Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005 > i686 athlon i386 GNU/Linux > > > > > > It didn't come with mpg123 so I downloaded it from the internet. MOH > works, but it is terribly loud and mistorted. Tried running under > quitemp3 profile but it didn't help. > > > > I feel like there is something I may be missing here. Any ideas??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8
Running Asterisk 1.2.8 on Pound Key linux which I downloaded from Digium site. Uname output: Linux localhost 2.6.13.4-1.x86.i686.cmov #1 Wed Nov 23 11:31:48 EST 2005 i686 athlon i386 GNU/Linux It didn’t come with mpg123 so I downloaded it from the internet. MOH works, but it is terribly loud and mistorted. Tried running under quitemp3 profile but it didn’t help. I feel like there is something I may be missing here. Any ideas??? Thanks in advance. Jeronimo. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == ___ --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
