[asterisk-users] ITSP's no longer supporting IAX?
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD ___ -- 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] ITSP's no longer supporting IAX?
I use simple port forwarding on an Linux firewall (iptables)...so that's not the issue. I was referring to IAX2 of course (IAX has be gone a long time I think)... Unlimitel is running * 1.4.x (and so am I)... I just can't understand IAX2 connections suddenly dropping (on one day) being protocol issues (if no one changed their * versions). Or is this how IAX2 fails? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dr. Michael J. Chudobiak Sent: March 25, 2009 9:29 AM To: Asterisk Users List Subject: Re: [asterisk-users] ITSP's no longer supporting IAX? The choice of router/NAT is critical though. Unlimitel recommended the SnapGear 560 to me, and it eliminated all the issues I was having with IAX going through my Sonicwall devices. Just another datapoint for you... Just curious. Since IAX only uses ONE port, do you have any idea what the technical reason behind a specific router would be critical? Well, with a Sonicwall TZ170, you had to enabled Firewall VOIP Enable consistent NAT, which was not the default setting. Then, you had to figure out that Firewall Advanced Default UDP Connection Timeout defaulted to 30 seconds, less than the normal Asterisk 60 second registration timeout. Then, for some reason, the TZ170 would simply lose packets. A fraction of calls would end almost immediately after they started, with Asterisk reporting a raw hangup error and INVAL packets, suggesting that some IAX2 packets were being lost, mis-ordered, or mis-translated. Anyway, the Sonicwall TZ170 was totally unreliable for IAX2 connections. They caused me a lot of grief. Avoid them like the plague. The Snapgear 560 just works, which I appreciate very much! - Mike ___ -- 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] How to verify availability of the DID connection?
Robert, We've helped clients setup monitoring scripts for this type of situation - 2 different ways. One is a ping script, the other monitors the asterisk peer status of registration. These were temporary until they could get to the root cause however. Since you have multiple providers going down, I would dig into the cause on your end... What diagnostics have you done so far? Michelle Dupuis www.generationd.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: March 7, 2009 9:36 AM To: Asterisk Users List Subject: Re: [asterisk-users] How to verify availability of the DID connection? All these questions are valid, though I want first to see that the DID does not work then I will go and try to resolve it. I do not have a specific issue at this moment. Sincerely, Robert Augustyn p:519.997.3106 ext:802 m:519.817.2503 www.linqone.com -Original Message- From: asterisk@sedwards.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, March 06, 2009 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to verify availability of the DID connection? On Thu, 5 Mar 2009, Robert Augustyn wrote: Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Are you losing DIDs that terminate on your Asterisk box or your clients Asterisk box? Are these DIDs registering with Asterisk and are you re-registering often enough? Is it a problem within the providers? Can you port the DIDs to another provider? Why do the DIDs stop working? Is is a connectivity problem you could detect with something like ping or Nagios? Since you say different providers I'm thinking a general connectivity problem or something generally out of whack with registrations. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] How to verify availability of the DID connection?
There's a defaultexpirey setting in sip.conf but I wouldn't go there yet. Does your ping work sometimes and not other times? Have you done route/network diagnostics? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: March 7, 2009 3:39 PM To: Asterisk Users List Subject: Re: [asterisk-users] How to verify availability of the DID connection? Importance: High Thanks, Well sometimes I have a situation that the trunk is registered but there is no communication coming in. So ping and looking for registration status does not work ... When I run sip reload it starts working again ? One difference is that I can see is the refresh on the registration is 585 and not the usual 105. Can I adjust this down anywhere? Sincerely, Robert Augustyn p:519.997.3106 ext:802 m:519.817.2503 www.linqone.com -Original Message- From: supp...@ocg.ca [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical Support Sent: Saturday, March 07, 2009 9:59 AM To: 'Asterisk Users List' Subject: Re: [asterisk-users] How to verify availability of the DID connection? Robert, We've helped clients setup monitoring scripts for this type of situation - 2 different ways. One is a ping script, the other monitors the asterisk peer status of registration. These were temporary until they could get to the root cause however. Since you have multiple providers going down, I would dig into the cause on your end... What diagnostics have you done so far? Michelle Dupuis www.generationd.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Augustyn Sent: March 7, 2009 9:36 AM To: Asterisk Users List Subject: Re: [asterisk-users] How to verify availability of the DID connection? All these questions are valid, though I want first to see that the DID does not work then I will go and try to resolve it. I do not have a specific issue at this moment. Sincerely, Robert Augustyn p:519.997.3106 ext:802 m:519.817.2503 www.linqone.com -Original Message- From: asterisk@sedwards.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, March 06, 2009 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to verify availability of the DID connection? On Thu, 5 Mar 2009, Robert Augustyn wrote: Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Are you losing DIDs that terminate on your Asterisk box or your clients Asterisk box? Are these DIDs registering with Asterisk and are you re-registering often enough? Is it a problem within the providers? Can you port the DIDs to another provider? Why do the DIDs stop working? Is is a connectivity problem you could detect with something like ping or Nagios? Since you say different providers I'm thinking a general connectivity problem or something generally out of whack with registrations. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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] $20 Bounty
Damn you for solving this before he upped the bounty by a pack of tictacs!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: March 3, 2009 10:51 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] $20 Bounty
On Tue, Mar 3, 2009 at 10:25 PM, David Backeberg dbackeb...@gmail.com
wrote:
exten = 123,s,1 Playback(enterzipcode)
exten = 123,s,n Read(zip||5)
exten = 123,s,n System(wget http://pathtoyahooservice${zip} -o
forecast.txt)
exten = 123,s,n System(wget --post-file forecast.txt -o wav.url)
exten = 123,s,n System(wget --input-file wav.url -o voice.wav)
exten = 123,s,n Playback(voice)
exten = 123,h,1 Hangup
On Tue, Mar 3, 2009 at 7:12 PM, Dean Collins d...@cognation.net wrote:
I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk
Weather App on Tropo.
All you have to do is violate the ToS on a few services:
wget the weather from yahoo, for instance:
http://weather.yahooapis.com/forecastrss?p=06513
Conditions for New Haven, CT at 9:53 pm EST
Current Conditions:
Fair, 20 F
Forecast:
Tue - Clear. High: 25 Low: 13
Wed - Mostly Sunny. High: 34 Low: 19
do a wget post of that output from the previous wget to
http://www.research.att.com/~ttsweb/tts/demo.php
do a wget on the wav file that demo generates.
It would be nicer if you record a prompt before asking for the
zipcode, but it's not strictly necessary.
You can paypal me the cash to my email. The legitimate license for
ATT Natural Voices is more than $20, and nothing built into Asterisk
for free is going to give you free-form text-to-speech.
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Re: [asterisk-users] How to generate core dump?
Install a Microsoft product. (Sorry I couldn't resist when I saw the subject) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: March 4, 2009 8:48 AM To: Asterisk Users List Subject: Re: [asterisk-users] How to generate core dump? On Wed, Mar 04, 2009 at 09:32:43AM +0100, Klaus Darilion wrote: Mark Michelson schrieb: Ken D'Ambrosio wrote: Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in make menuconfig and didn't see anything appropriate. Thanks, -Ken Run Asterisk with the -g option and it will dump a core file if it should crash. If you also want to specify the location/file name this can be useful too (man core) echo /tmp/core.%p /proc/sys/kernel/core_pattern Hmm.. this way you can't tell which executable generated it . echo /tmp/core.%e.%t /proc/sys/kernel/core_pattern Or maybe (untested) echo |/usr/local/sbin/core_handler '%e' '%s' See the kernel documentation: http://kernel.org/doc/Documentation/sysctl/kernel.txt This is handy for those of you with limited disk space. OTOH, it will probably not work on legacy systems with kernel 2.6.18. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] $20 Bounty
Perhaps if he threw in a paperclip and some tictacs people would respond... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: March 3, 2009 7:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] $20 Bounty On Tue, 3 Mar 2009, Dean Collins wrote: I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk Weather App on Tropo. Wow. $20. cricketcricketcricket :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Windows Mobile MWI and asterisk
Has anyone written a Windows Mobile app which gets the MWI info from a SIP server, and updates the VM counter in the OS? I'd like my PPC to show my voicemail count (and SIP MWI seems like the easiest way) ___ -- 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] strange text message:)
Are you sure this is not just a standard SIP MWI message? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S. Sent: February 23, 2009 8:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] strange text message:) is any chance to use this feature to send messages on this kind of phones? On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote: you are getting the info about the voicemail becausethe soft on your phone support it. in sip.conf you can find some parameters to send that info. in other soft phones like x-lite you will have the same info. David 2009/2/23 Catalin S. jonsonpla...@gmail.com Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested about how can I send these messages with other information's or whatever I want to my terminals. Also I observed that works with Nokia E71 only. Maybe is because I updated some software on It , Not Firmware. Do you guys observed this too? Thank you for support. Catalin. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Pingable and Unreachable at the same time !
Did you use the same screen name / name for the 2 SIP extensions you setup on the one phone? If so, some phones will confuse asterisk based on the SIP header (in particular AASTRA phones). If this is an Aastra phone, this is probably the cause... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: February 17, 2009 8:47 AM To: Asterisk Users List Subject: Re: [asterisk-users] Pingable and Unreachable at the same time ! 2009/2/17 Marc STORCK msto...@voipgate.com Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Yes. I think that simply, in this case, the endpoint (SIP phone) is just broken : it wouldn't reply to anything ... I'm not 100% sure now, but wouldn't be surprised ... Regards, Marc From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: mardi 17 février 2009 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Strange dialplan matching issue
No, sorry, we match _XXX to jump to plant123
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: February 13, 2009 4:35 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange dialplan matching issue
Importance: High
OCG Technical Support schrieb:
We use extensions like plant201 and tunnel12 so it does work in 1.4
As a *pattern* (e.g. _plant2XX, _tunnel.)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
Espinal
On extensions.conf.sample I see this:
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
Maybe after using '_' Asterisk is waiting for one of the above pattern
matching characters.
a. The 'hilton-' part of your dialplan might not being considered valid,
and Asterisk *might* be trying to match the 'XX' part LITERALLY, and
would be trying to reach extension '2XX'
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
b. then, in:
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
You provided the real extension number (after you take out the fist 7
digits).
So, Asterisk reaches '203', etc.
Try only using valid pattern matching characters in your dialplan to see
if it works.
Chris Bagnall wrote:
Wondering if anyone has come across this strange dialplan pattern
matching
issue before:
I have a context defined as follows (the plus simply implies it follows
on
from an existing context in another #include - which, yes, has been
included
first):
[privatedundi](+)
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt
from ip masked, request 'hilton-...@privatedundi' does not exist
Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
Dialling hilton-202 now works every time.
The *really* strange thing is that I have a number of similar pattern
matches, and all the others work fine, it's just this one that doesn't.
The box in question is running 1.4.22, but I have had a similar issue in
the past with a 1.2 box, so it does not appear to be version specific.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Strange dialplan matching issue
We use extensions like plant201 and tunnel12 so it does work in 1.4
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: February 11, 2009 10:16 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange dialplan matching issue
On extensions.conf.sample I see this:
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
Maybe after using '_' Asterisk is waiting for one of the above pattern
matching characters.
a. The 'hilton-' part of your dialplan might not being considered valid,
and Asterisk *might* be trying to match the 'XX' part LITERALLY, and
would be trying to reach extension '2XX'
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
b. then, in:
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
You provided the real extension number (after you take out the fist 7
digits).
So, Asterisk reaches '203', etc.
Try only using valid pattern matching characters in your dialplan to see
if it works.
Chris Bagnall wrote:
Greetings list,
Wondering if anyone has come across this strange dialplan pattern matching
issue before:
I have a context defined as follows (the plus simply implies it follows on
from an existing context in another #include - which, yes, has been included
first):
[privatedundi](+)
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt
from ip masked, request 'hilton-...@privatedundi' does not exist
Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
Dialling hilton-202 now works every time.
The *really* strange thing is that I have a number of similar pattern
matches, and all the others work fine, it's just this one that doesn't.
The box in question is running 1.4.22, but I have had a similar issue in
the past with a 1.2 box, so it does not appear to be version specific.
Any thoughts?
TIA.
Regards,
Chris
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--
Jose P. Espinal
http://www.eSlackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs
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Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6
Don't expect too much from Aastra. In our previous dealings trying to report serious bugs (like phone lockup/crash) to Aastra, they didn't want the details, or they simply gave us canned answers which did no good. (Superficial tech support) We've moved away from Aastra for new installs, but we still have to support old customers with Aastra -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: February 11, 2009 12:45 AM To: Asterisk Users List Subject: Re: [asterisk-users] Aastra phone crashes with Asterisk 1.6 Carlos Chavez schrieb: I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. If I dial any of those phones the call will drop after a minute or so and the phone will crash. I'm not saying it's not an Asterisk problem. Maybe something in the SIP signaling/RTP is broken. However it's definitely an Aastra problem. No matter how broken the signaling -- that's no excuse for crashing. So make sure to report the issue to Aastra as well. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Contact lookup
Have a look at smartCID at www.generationd.com
Uses a simple mySQL database, allows for call blocking flag, reverse CID
lookup, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: February 3, 2009 11:51 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Contact lookup
On Tue, 3 Feb 2009, Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
I'd like to be able to look up each incoming CLI to retrieve an
associated name, if available, and then pass that to the extensions so
that they can see both the name and number of the caller. I'm not
after LDAP or anything else maintained externally, just a contact
lookup for my system.
I suspect that Astdb could be used for this, as could a relational
database like MySQL or postgres (accessed via AGI?) Probably simpler
would be to maintain a text configuration file since I'm only
concerned about less than a hundred entries initially.
I'd appreciate insight into which is the easiest way to do this, and
also any pointers to tutorials etc.
AstDB:
At it's very simplest:
exten = s,n,Set(CALLERID(name)=Unknown)
exten = s,n,Set(name=${DB(cid/${CALLERID(number)})})
exten = s,n,GotoIf($[${name} = ]?endCID)
exten = s,n,Set(CALLERID(name)=${name})
exten = s,n(endCID),Noop(fixCallerID - End of processing - returning
${CALLERID(all)})
... somewhere in the incoming processing. (This is an extract from an
overly complcated macro I use) Things to check for - a name already being
present - eg. on an incoming SIP call. No name in the astDB - might want
to substitute Unknown ..
All you need to do now is populate the astDB - I use a web interface and
some php to drive the manager interface...
My biggest site has just under 300 lookup entries... (Which presents other
issues with the web interface, but ...)
Gordon
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Re: [asterisk-users] Quiet 24 port POE gig switch
Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- 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] Quiet 24 port POE gig switch
My google search says fanless... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales Sent: February 1, 2009 6:49 PM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch Importance: High My memory of a HP procurve (a 2626 PWR from memory) was that it was quite noisy - have they changed? PaulH OCG Technical Support wrote: Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- 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 ___ -- 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] Quiet 24 port POE gig switch
A little off topic but I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? 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
[asterisk-users] vmail.cgi - permissions error help
We always install native Asterisk (not when over the other packaged versions) I tried setting the SUID bit on the vmail.cgi file but that didn't help...so I must be missing something. Can someone else suggest a fix? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gondar Monn Sent: January 31, 2009 2:17 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Importance: High Have you tried FreePBX ? It allows Asterisk administration via web interface and has module just for that. G. On Fri, Jan 30, 2009 at 7:56 PM, OCG Technical Support supp...@ocg.ca wrote: No - the server generates the error: Software error: Hrm, can't seem to open /var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav For help, please send mail to th From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 10:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Importance: High It might be browser security issues? Have you tried with different browsers? On Fri, Jan 30, 2009 at 9:53 PM, OCG Technical Support supp...@ocg.ca wrote: Strangely, I can DELETE the messages from the web interface...just playback causes a permission error... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical Support Sent: January 30, 2009 9:40 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? I just tried the vmail.cgi app. Although working, there is clearly a permissions problem preventing playing the wav files. I run Fedora 8, and the patch files (on the wiki) are apparently broken. Does anyone have a solution for fedora? Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 7:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Thank you Lenz. That's exactly what I'm looking for. On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We use the default web interface, it comes with a file called vmail.cgi in the defaiult Asterisk tree. Thanks l. 2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports Web interface for checking of voicemail. Does anyone know where I can find more information about this Web interface for checking of voicemail feature? If there is no such thing, what an alternative is? Do you think writing my own AGI script is the right way to go? Is there already an existing AGI scripts or other things that I should install on top of Asterisk that already do something like this? Basically, I try to avoid writing a bunch of codes to retrieve voicemail from my email and build a brand new voicemail web page. If there are something I can reuse, I'd like to use them. Thank you for your help. ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- 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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use
Re: [asterisk-users] Quiet 24 port POE gig switch
I can find FANLESS 24 port PoE 10/100, or FANLESS 24 port non-POE 10/100/1000 I guess I'll just have to wait for newer chips..till then dropping down to 10/100 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: January 31, 2009 10:02 PM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch Importance: High OCG Technical Support wrote: A little off topic but I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Chances of finding a PoE switch that is quiet out of the box is about as good as finding a government 'worker'. It's kind of an oxymoron. Of the switches I've used, the Linksys/Cisco line was the loudest. Dlink's were quieter, but still not something you'd want sitting next to a desk. About the only fanless PoE switches I've seen are the smaller Netgear's, but they are not Gigabit. Darrick ___ -- 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] Asterisk VoiceMail: Is there a web interface for checking voicemail?
I just tried the vmail.cgi app. Although working, there is clearly a permissions problem preventing playing the wav files. I run Fedora 8, and the patch files (on the wiki) are apparently broken. Does anyone have a solution for fedora? Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 7:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Thank you Lenz. That's exactly what I'm looking for. On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We use the default web interface, it comes with a file called vmail.cgi in the defaiult Asterisk tree. Thanks l. 2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports Web interface for checking of voicemail. Does anyone know where I can find more information about this Web interface for checking of voicemail feature? If there is no such thing, what an alternative is? Do you think writing my own AGI script is the right way to go? Is there already an existing AGI scripts or other things that I should install on top of Asterisk that already do something like this? Basically, I try to avoid writing a bunch of codes to retrieve voicemail from my email and build a brand new voicemail web page. If there are something I can reuse, I'd like to use them. Thank you for your help. ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- 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] Asterisk VoiceMail: Is there a web interface for checking voicemail?
Strangely, I can DELETE the messages from the web interface...just playback causes a permission error... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical Support Sent: January 30, 2009 9:40 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? I just tried the vmail.cgi app. Although working, there is clearly a permissions problem preventing playing the wav files. I run Fedora 8, and the patch files (on the wiki) are apparently broken. Does anyone have a solution for fedora? Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 7:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Thank you Lenz. That's exactly what I'm looking for. On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We use the default web interface, it comes with a file called vmail.cgi in the defaiult Asterisk tree. Thanks l. 2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports Web interface for checking of voicemail. Does anyone know where I can find more information about this Web interface for checking of voicemail feature? If there is no such thing, what an alternative is? Do you think writing my own AGI script is the right way to go? Is there already an existing AGI scripts or other things that I should install on top of Asterisk that already do something like this? Basically, I try to avoid writing a bunch of codes to retrieve voicemail from my email and build a brand new voicemail web page. If there are something I can reuse, I'd like to use them. Thank you for your help. ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- 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] Asterisk VoiceMail: Is there a web interface for checking voicemail?
No - the server generates the error: Software error: Hrm, can't seem to open /var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav For help, please send mail to th From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 10:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Importance: High It might be browser security issues? Have you tried with different browsers? On Fri, Jan 30, 2009 at 9:53 PM, OCG Technical Support supp...@ocg.ca wrote: Strangely, I can DELETE the messages from the web interface...just playback causes a permission error... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical Support Sent: January 30, 2009 9:40 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? I just tried the vmail.cgi app. Although working, there is clearly a permissions problem preventing playing the wav files. I run Fedora 8, and the patch files (on the wiki) are apparently broken. Does anyone have a solution for fedora? Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 7:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Thank you Lenz. That's exactly what I'm looking for. On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We use the default web interface, it comes with a file called vmail.cgi in the defaiult Asterisk tree. Thanks l. 2009/1/30 Soonthorn Ativanichayaphong soonth...@yapinc.com Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports Web interface for checking of voicemail. Does anyone know where I can find more information about this Web interface for checking of voicemail feature? If there is no such thing, what an alternative is? Do you think writing my own AGI script is the right way to go? Is there already an existing AGI scripts or other things that I should install on top of Asterisk that already do something like this? Basically, I try to avoid writing a bunch of codes to retrieve voicemail from my email and build a brand new voicemail web page. If there are something I can reuse, I'd like to use them. Thank you for your help. ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- 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 -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- 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] Looking for SIP loud ringer
Connect an amp to the onboard speaker (run wires out the case)...then you can really blast the ring! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: January 28, 2009 10:53 AM To: Asterisk Users List Subject: Re: [asterisk-users] Looking for SIP loud ringer Danny, Thanks for the idea, I thought of it but I was looking for a more elegant solution, and one that would as much as possible not require my intervention in any way. A PC requires support even in the best of times: it`s got harddrives, software, patches, etc, etc. An alternative would be a SIP phone with a very loud max ring, but that`s not the case with the phones I know (Polycoms) Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, January 28, 2009 10:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Looking for SIP loud ringer Why don't you put a PC in the storeroom with a softphone to be the loud ringer? You could make the ring though the speakers be as loud as the system would support. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, January 28, 2009 9:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Looking for SIP loud ringer Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike ___ -- 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] Stutter/chopoff first audio played
I've noticed on a few installations that the very first audio played after a call in answered (eg: Greeting), the first part of the audio is cutoff/stuttered. Is this because Asterisk needs some RTP to create a sync for audio - and the first 1 second is lost? Should one play 1 sec of silence first? ___ -- 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] Using a sidecar? Ideas?
I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)...but what else? Anyone want to share interesting things they've done with a sidecar? ___ -- 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] Using a sidecar? Ideas?
Interesting...could you give some details on: Out of office mode: How would BLF be used for that? Are you using the sidecar button as a light with ON/OFF status? How do you do that with asterisk / your sidecar button? (BLF just subscribes to a device in use I thought) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: January 18, 2009 3:11 PM To: Asterisk Users List Subject: Re: [asterisk-users] Using a sidecar? Ideas? Importance: High OCG Technical Support schrieb: I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)... For receptionists BLFs are a necessity. but what else? Anyone want to share interesting things they've done with a sidecar? People love things like using BLF for night mode / out of office mode, queue login state, ... Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Using a sidecar? Ideas?
For these features, are you just sending a series of DTMF's (like speed dial)? Or are you somehow trigger the lamp on/off too? If so, how do you do that? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: January 18, 2009 9:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Using a sidecar? Ideas? Importance: High Here is a list of stuff that I can remember: BLF Group login/logout Day/Night mode Call Record Speed dial On Sun, Jan 18, 2009 at 2:43 PM, OCG Technical Support supp...@ocg.ca wrote: I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)...but what else? Anyone want to share interesting things they've done with a sidecar? ___ -- 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] how to debug mime-construct with fax2mail?
If you want to email me your fixed script I'll put it up on the web site... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: January 15, 2009 7:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] how to debug mime-construct with fax2mail? Lyle Giese wrote: If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle You are the man. If we ever meet I owe you a beer, at least one. In the fax2mail script, it just calls mime-construct without a full path. mime-construct on my box is in /usr/local/bin which must not be in the path of the environment System calls are run in. Putting in the fully qualified path made it work. Thanks again. sean ___ -- 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] 1.6.1-b4: Can't get fax2mail work from System()
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 14, 2009 5:31 PM
To: Asterisk Users List
Subject: [asterisk-users] 1.6.1-b4: Can't get fax2mail work from System()
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0$
{CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email ${Sean_email} -f ${FAXFILE})
which looks like it works just fine from the cli:
-- DAHDI/2-1 is ringing
-- Redirecting DAHDI/4-1 to fax extension
-- Hungup 'DAHDI/2-1'
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
-- Goto (incoming-fax,s,1)
-- Executing [...@incoming-fax:1] Set(DAHDI/4-1,
FAXFILE=/var/spool/asterisk/fax/200901141711-0) in new stack
-- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1,
/var/spool/asterisk/fax/200901141711-0.tif) in new stack
-- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack
== Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1'
-- Executing [...@incoming-fax:1] System(DAHDI/4-1,
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0) in
new stack
-- Hungup 'DAHDI/4-1'
But it doesn't - no email is ever sent. BUT, if I execute the fax2mail
cmd from the terminal (pasting from the cli output) it sends the email:
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0
Am I screwing up the System() command somehow? Is System() screwed up in
1.6.1?
Any clues how to debug this? I did find one relevant thread
http://asteriskforum.ru/viewtopic.php?p=15629 , which is unfortunatley
in Russian. In that thread someone figured out how to turn on DEBUG for
app_fax. How did you do that?
sean
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Re: [asterisk-users] recommendation for German sound files
I can do a great Colonel Klink and pretty good Shulz imitation...in case you want me to record some prompts. :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion Sent: January 7, 2009 11:31 AM To: Asterisk Users List Subject: [asterisk-users] recommendation for German sound files Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+internatio nal#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus ___ -- 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] Using DECT phones as SIP phones?
I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think) ___ -- 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] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call
I had front speaker working initially - but have lost that (now back only). Something isn't quite right - but still workable... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: December 4, 2008 3:10 AM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call Importance: High From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Wednesday, December 03, 2008 11:14 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call You’ll have to recheck your facts...MS does include a SIP client in WM6. And it works great ☺ Carriers/brands can remove items from ROM, but the SIP client is in by default. Have a look on XDA developers web site for details Jason, here’s what you need: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windows-mobile-6x-for-free-voip-calls-using-asterisk/ OCG; Have you managed to get this working on the front speaker? Or still the back speaker only? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.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
Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
I'm using the Wm6 built in client. (Enabled via CAB file to add-back files removed from ROM) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta Sent: December 3, 2008 8:42 PM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call What sip client are you using on WM6 side ? ___ -- 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] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call
You'll have to recheck your facts...MS does include a SIP client in WM6. And it works great J Carriers/brands can remove items from ROM, but the SIP client is in by default. Have a look on XDA developers web site for details From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Aarons (US) Sent: December 3, 2008 9:44 PM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call Importance: High Microsoft doesn't make a native SIP client in Windows Mobile you can use for a phone call. Do you mean Windows Live Messenger? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Wednesday, December 03, 2008 9:15 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't match request NOTIFY to call I'm using the Wm6 built in client. (Enabled via CAB file to add-back files removed from ROM) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hakem Ta Sent: December 3, 2008 8:42 PM To: Asterisk Users List Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call What sip client are you using on WM6 side ? _ Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client. Overall things are working well. However, I regularly get the following message: [Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. From what I've read, the client doesn't subscribe to MWI but gets a notify event - which it rejects. The voicemail notifications ARE working on the device. Any way to get rid of this message (while keeping the MWI on the phone)? ___ -- 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] OT: ATA causes random DTMF in stream
I've got a user with Linksys ATA's for their analog phones. At random times during calls, the other party hears DTMF tones during the call. Is there a way to solve this? ___ -- 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] set(CALLERID(name) not working
Take a look at smartCID at www.generationd.com
This tool will set callerid based on number in a database. If not listed
there, it will search 411 for reverse lookup etc.
It will also let you flag calls for blocking, etc..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: November 9, 2008 2:30 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] set(CALLERID(name) not working
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed:
Perhaps try this:
[set-callerid-name]
exten = 02025462677,1,Set(CALLERID(name)=Fred)
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Re: [asterisk-users] Sendmail using SMTP authorization
Could you send a link to the post you referenced? I'd like to get sendmail working with rogers too... Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude Sent: November 8, 2008 8:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] Sendmail using SMTP authorization Fixed thanks to Tilman's post on comp.mail.sendmail. I had smtp.broadband.rogers.com in authinfo, the same as what I had for SMART_HOST in sendmail.mc but I had to change authinfo to smtp-rog.mail.yahoo.com . I wasn't worried about this at first because a dig on smtp.broadband.rogers.com shows that it resolves to smtp-rog.mail.yahoo.com (Rogers uses Yahoo's infrastructure) so as far as I was concerned it was the same thing. In fact sendmail ends up trying to deliver the mail to the right place, but beacuse smtp-rog.mail.yahoo.com cannot be found in authinfo, the credentials cannot be found. Hope this helps someone else! Thanks for checking my post Matt. H On Tue, Nov 4, 2008 at 7:06 PM, Matt Gibson [EMAIL PROTECTED] wrote: Try using SSMTP http://www.linux.com/articles/132006 It works with any provider for mail sending, and takes 30 seconds to setup. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of hugolivude Sent: Tuesday, November 04, 2008 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Sendmail using SMTP authorization Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getting it to work. I've included the following below: Asterisk 1.4.21 CentOS 5 Sendmail 8.13.8 === bounced mail === === maillog === === hosts === === access === === authinfo === === sendmail.mc === The bounced mail file shows the authentication problem, although there's also a troubling DSN: Service unavailable message that appears in maillog. I'm not sure whether the two are related or if the latter is really a problem at all. Any help would be welcome. Thanks in advance! Cheers, Hugh CentOS 5 Sendmail 8.13.8 === bounced mail === = From [EMAIL PROTECTED] Sun Nov 2 11:53:57 2008 Return-Path: [EMAIL PROTECTED] Received: from localhost (localhost) by rapperyo.com (8.13.8/8.13.8) id mA2Gru4B002917; Sun, 2 Nov 2008 11:53:56 -0500 Date: Sun, 2 Nov 2008 11:53:56 -0500 From: Mail Delivery Subsystem [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: multipart/report; report-type=delivery-status; boundary=mA2Gru4B002917.1225644836/rapperyo.com Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --mA2Gru4B002917.1225644836/rapperyo.com The original message was received at Sun, 2 Nov 2008 11:53:56 -0500 from rapperyo.com [127.0.0.1] - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html) - Transcript of session follows - ... while talking to smtp-rog.mail.yahoo.com.: MAIL From:[EMAIL PROTECTED] 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html 554 5.0.0 Service unavailable --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/delivery-status Reporting-MTA: dns; rapperyo.com Received-From-MTA: DNS; rapperyo.com Arrival-Date: Sun, 2 Nov 2008 11:53:56 -0500 Final-Recipient: RFC822; [EMAIL PROTECTED] Action: failed Status: 5.0.0 Diagnostic-Code: SMTP; 530 authentication required - for help go to http://help.yahoo.com/help/us/mail/pop/pop-11.html Last-Attempt-Date: Sun, 2 Nov 2008 11:53:56 -0500 --mA2Gru4B002917.1225644836/rapperyo.com Content-Type: message/rfc822 Return-Path: [EMAIL PROTECTED] Received: from rapperyo.com (rapperyo.com [127.0.0.1]) by rapperyo.com (8.13.8/8.13.8) with ESMTP id mA2Gru4B002915 for [EMAIL PROTECTED]; Sun, 2 Nov 2008 11:53:56 -0500 Received: (from [EMAIL PROTECTED]) by rapperyo.com (8.13.8/8.13.8/Submit) id mA2GrtoD002914; Sun, 2 Nov 2008 11:53:55 -0500 Date: Sun, 2 Nov 2008 11:53:55 -0500 From: root [EMAIL PROTECTED] Message-Id: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: I'm sending mail from the Terminal! --mA2Gru4B002917.1225644836/rapperyo.com-- === maillog === Nov 2 11:49:35 pbx sendmail[2421]: alias database /etc/aliases rebuilt by root Nov 2 11:49:35 pbx sendmail[2421]: /etc/aliases: 76 aliases,
[asterisk-users] VoIP traffic shaping
This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- == Thanks Kristian I will checkout the new script and see how it goes! Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Friday, October 31, 2008 1:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] fax / t38 gateway On 10/31/08, Jonn R Taylor [EMAIL PROTECTED] wrote: Here is the QOS script that I use on my bridge. http://www.taylortelephone.com/asterisk/astshape You should upgrade to the newer astshape script. It classifies traffic using iptables, which is much more flexible. It also has beta support for the HFSC qdisc: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/package/iproute2/a stshape -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
Re: [asterisk-users] XML Cisco config file
Post it on the wiki! Im sure Ill need it someday From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César García Sent: October 29, 2008 6:54 PM To: Asterisk Users List Subject: Re: [asterisk-users] XML Cisco config file Well guys I got it, I started up again making the xml file according to this: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configurati on+files+for+SIP#Downgradingthefirmware And... voila ! 7911G working with Asterisk and firmware 8.4.0!!! if anybody need the xml, let me know :) 2008/10/28 Lincoln King-Cliby [EMAIL PROTECTED] I'm not sure if it's the only issue but you're going to have issues with phonelabelEtiqueta_del_telefono/phonelabel The text within the phonelabel tag is a maximum of 11 or 12 characters (I can't remember off the top of my head), if it's longer than that--I count 21 characters in the example, the phone will reject the entire configuration file more or less silently (it is logged in the phone's debug log at http://phone ip address/ but there's no display on the phone itself). That sounds like at least part of what's happening in your case. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.controlworks.com/ http://www.controlworks.com Crestron Authorized Independent Programmer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César García Sent: Tuesday, October 28, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] XML Cisco config file Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done thanks a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.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 -- http://celord.blogspot.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
Re: [asterisk-users] Sporadic One Way Audio
How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] Sporadic One Way Audio
Well, if this is snom specific I can't offer more insight. It really sounds like misconfigured iptables and/or sip helper (conntrack/nat/etc). Are you sure your IP address is right in your sip.conf? If you don;t have NAT set to yes for these phones, they will trust the sip header for IP address and may misroute. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 7:36 PM To: Asterisk Users List Subject: Re: [asterisk-users] Sporadic One Way Audio Importance: High The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card. The lost RTP would have be between the Asterisk server and the phones. There are only 2 phones in the building, 2 lines coming in to the asterisk server and the server is on the same ethernet switch as the phones. The phones are SIP phones. This is a simple PBX system that picks up calls from the analog lines and routes them to the appropriate phone, although it will eventually be linked to a larger system once all the minor bugs are resolved. OCG Technical Support wrote: How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1 out? Etc... Are you looking for lost RTP between * and internal phones or * and external provider? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: October 24, 2008 5:55 PM To: Asterisk Users List Subject: [asterisk-users] Sporadic One Way Audio I'm having an unusual problem at one of my branch offices. Every now and then they will make a call and the person they call is unable to hear them, but they are able to hear the person. The Asterisk server has only one ethernet interface and is on the same physical network as the 2 snom 300 phones and is connected to the PSTN lines with a Rhino R4FXO-EC card. Usually hanging up and calling back solves the problem, but it is still aggravating to the customer that has been called. Normally I'd suspect that something was only passing packets in one direction, but there is no firewall between the asterisk server and the phones and no iptables or anything like that running on the Asterisk server and sifting through sip debug logs to try to find one call out of maybe 50 has so far proven fruitless. Are there any common issues that might cause this? Thanks, Brent Davidson ___ -- 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] conntrack_sip, iptables, and asterisk
I have a new Fedora 9 firewall I am setting up in front of an Asterisk 1.4 box. I ported over all of my iptables rules..but now have a strange problem: SOMETIMES, the audio is only 1-way (i.e. and RTP path problem). Can someone offer a tip here? Since I have conntrack_sip loaded on the firewall, do I need to: 1. Use SIP and RTP port forwarding prerouting to my asterisk box? (SIP clients are outside the LAN) - this is the way I do it now 2. Remove all SIP and RTP port forwarding prerouting and assume conntrack_sip will do everything? 3. Allow SIP and RTP *INTO* the firewall, to allow conntrack_sip to work? Clearly something has changed with conntrack_sip or iptables in the latest kernel...so I need to figure this out. Help! Thanks! Michelle ___ -- 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] SCCP port numbers used for audio stram?
I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SCCP - max lines per phone limit
I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 Is there a setting to tell Asterisk how many lines to permit per phone? (The 7921 should allow for 6 lines according to the manual) Thanks MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setup speed dials on Cisco 7921
I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setup speed dials on Cisco 7921
Chan_sccp again... From what I read chan_sccp is the successor to chan_skinny. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which internet phone protocol best to choose
I would have said the short answer is IAX :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: September 12, 2008 7:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Which internet phone protocol best to choose The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan help.
This is some pretty basic stuff... (someone will probably send you a RTFM) Start with the sample dialplan (make samples I think)...trace the dialplan along to understand how it works Check the wiki and then post anything that you need help with From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sydney Web Hosting Sent: July 6, 2008 8:33 PM To: Asterisk Users List Subject: [asterisk-users] dial plan help. I have a question about the following dial plan. Ring main number playback message If press 1 got to support if press 2 go to sales //Support Play message your call is important to us then ring the phone and I pickup. //Sales Play message your call is important to us then ring the phone and I pickup. but, the problem is I only have 1 staff member at the moment. So how do we set it up if I'm out of the office, or on the mobile phone and can't answer the call. How does it know to go to voice mail? Regards Jared ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
I little more digging and I confirmed that cell phone VM and FAX waiting icons are in fact controlled by a proprietary SMS message format. Here's what I found: Message Waiting Indication Group: Store Message This Group allows an indication to be provided to the user about the status of types of message waiting on systems connected to the GSM PLMN. The mobile may present this indication as an icon on the screen, or other MMI indication. The mobile may take note of the Origination Address for messages in this group and group 1100. For each indication supported, the mobile may provide storage for the Origination Address which is to control the mobile indicator. Text included in the user data is coded in the Default Alphabet. Where a message is received with bits 7..4 set to 1101, the mobile shall store the text of the SMS message in addition to setting the indication. Bits 3 indicates Indication Sense: Bit 3 0 Set Indication Inactive 1 Set Indication Active Bit 2 is reserved, and set to 0 Bit 1 Bit 0 Indication Type: 0 0 Voicemail Message Waiting 0 1 Fax Message Waiting 1 0 Electronic Mail Message Waiting 1 1 Other Message Waiting* * Mobile manufacturers may implement the Other Message Waiting indication as an additional indication without specifying the meaning. The meaning of this indication is intended to be standardized in the future, so Operators should not make use of this indication until the standard for this indication is finalized. Now the tough part...does anyone want to create an app to send notification to a cell phone to set/clear these bits? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Send cell phone #VM waiting, just like cell carrier
We have a number of clients who have replaced their cell carrier voicemail with Asterisk (call forward no answer to * box). One feature they miss is that the cell carriers send the phone a message showing # voicemails waiting. Can Asterisk do the same somehow? MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier
Well, I realize that there must be some proprietary protocol between the carrier and the phone, since they have a dedicate spot on the cell screen for # VM waiting... As for an SMS message, is there a module/app which allows easy SMS messaging? (I looked a couple of years ago but only found commercial modules) MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 22, 2008 11:28 AM To: Asterisk Users List Subject: Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier On Sun, Jun 22, 2008 at 11:23 AM, OCG Technical Support [EMAIL PROTECTED] wrote: We have a number of clients who have replaced their cell carrier voicemail with Asterisk (call forward no answer to * box). One feature they miss is that the cell carriers send the phone a message showing # voicemails waiting. Can Asterisk do the same somehow? MD You could easily send them an SMS telling them they have X number of new messages and X number of saved messages. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier
We already do thatbut: If users turn their phone on after being out of range/server/power off, the carrier sends a VM notification. Also, not all phones have POP client capabilities... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: June 22, 2008 11:43 AM To: Asterisk Users List Subject: Re: [asterisk-users] Send cell phone #VM waiting, just like cell carrier I just send an email with the voicemail message details. Cool part about this is the attachment can be downloaded and played on the phone via windows media player. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, 22 June 2008 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Send cell phone #VM waiting,just like cell carrier On Sun, Jun 22, 2008 at 11:23 AM, OCG Technical Support [EMAIL PROTECTED] wrote: We have a number of clients who have replaced their cell carrier voicemail with Asterisk (call forward no answer to * box). One feature they miss is that the cell carriers send the phone a message showing # voicemails waiting. Can Asterisk do the same somehow? MD You could easily send them an SMS telling them they have X number of new messages and X number of saved messages. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mapping multimedia keys: pressed key not recognized
Wrong listsorry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: June 19, 2008 4:38 AM To: Asterisk Users List Subject: Re: [asterisk-users] Mapping multimedia keys: pressed key not recognized On Wed, Jun 18, 2008 at 08:21:06PM -0400, OCG Technical Support wrote: I've tried a few approaches to making the multimedia keys on my kbd play nice with myth, but all have lead to dead ends. One such dead end is to post this question to the Asteris Users mailing list, I guess :-( Wrong list? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mapping multimedia keys: pressed key not recognized
I've tried a few approaches to making the multimedia keys on my kbd play nice with myth, but all have lead to dead ends. I decided to take the simple approach, and use the myth setup menu for keyboard mappings. Now, I have myth (0.20) waiting for a key with Press a key, but when I press the PLAY button on my keyboard, myth says pressed key not recognized. How do I get myth to recognize the multimedia keys? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on SLOW solid state disk
I'm looking at building up a standard asterisk system fanless/no moving parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it is SLOW...25mb/sec read 8mb/sec write. Has anyone tried a slow disk like this on asterisk? Will this delay voice prompts or screw up ast/linux in any interesting way? (I know there are linux distros and Asterisk projects designed to run off CF, but I'm hoping to stay mainstream) Thanks, MD ___ -- 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] Interoffice phone setup
Change the order of resolution (hosts first, then DNS) and add relevant entries to your hosts table. That makes asterisk happy w/o an internet connection. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: June 9, 2008 9:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Interoffice phone setup On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote: On June 9, 2008 07:49:13 pm Joseph L. Casale wrote: What type of PBX hardware do you have on-site? Also what make/models of phones? Michael/Darryl, I do have a local asterisk box, which is why I am baffled. I am new to Asterisk and there is lots to learn, but my config is pretty basic, my sip.conf simply has the phones and single sip provider context in it. It doesn't make sense that the voip provider going offline takes the whole setup out with it. I am suspecting something else went south at the same time. I have snom m3's and one Astra 480i. Thanks! jlc I've seen this behaviour from Asterisk as well... while I can't say I have tracked it down and verified this... I've seen other talks about how Asterisk gets rather unhappy when it can't preform DNS queries. I suspect that may be your problem. Might want to check the archives for other issues that people have talked about DNS as a possible cause and see if there are any similarities. Yes, this is very true. Asterisk gets backed up trying to deal with lack of DNS. I'd diable the SIP trunk then restart. Perhaps this would permit internal calls to resume, as long as there are no attempts to dial external numbers. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Logitech DiNovo Mini keyboard with myth
Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- 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] Logitech DiNovo Mini keyboard with myth
I found the necessary keyboard codes and created a mapping in .Xmodmap, and then finally: /usr/bin/xmodmap $HOME/.Xmodmap Still, myth doesn't seem to care about the new keysnow what? How do I make myth map these new codes to myth actions? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: June 6, 2008 9:03 PM To: Asterisk Users List Subject: [asterisk-users] Logitech DiNovo Mini keyboard with myth Has anyone create the necessary config/kbd file to allow the DiNovo mini to work well with myth? (Mapped all of the multimedia buttons etc) =MD= ___ -- 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] Fedora 9 + Asterisk
Anyone tried Asterisk with Fedora 9 (recent release)? ___ -- 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] voicemail not sending emails
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
To: Asterisk Users List
Subject: [asterisk-users] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
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Re: [asterisk-users] VOICEMAIL OPTIONS help needed
We also have a script available (on www.generationd.com) which allows a user to reply to an emailed voicemail, which then deletes the associated VM file on the asterisk box. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: May 7, 2008 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] VOICEMAIL OPTIONS help needed see voicemail.conf.sample all the options you need are documented there. maxmsg delete On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote: Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. ___ -- 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] Storing voicemail on samba share
A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. We're thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? Thanks, MD ___ -- 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] Delete voicemail messages on asterisk by replying to email
Like many users I get my voicemails emailed to me, AND left on the asterisk server, so that I can retrieve them by phone or by email. However, I was frustrated that after I deleted a message in outlook that I still had to delete it from asterisk manually. So, I wrote a script that runs on the asterisk box that allows the user to simply reply to an email (from the asterisk pbx) - which causes the associated message files on asterisk to be deleted. It's a safe script in that it will NOT delete a message if there is any ambiguity as to which voicemail message files match the email, etc. It also has full logging (in /var/log/asterisk/voicemailcontrol). Email me if you want to try it. (You will have to rename the voicemail directory etc since it's hard coded into the script. I have not created a config file yet...) -MD- ___ -- 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] Delete voicemail messages on asterisk by replying to email
After lots of interest I've stopped emailing people the script and have made it available for download from www.generationd.com Look in the Downloads | Asterisk section. Be sure to read the readme AND the top of the script for instructions... ___ -- 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] Automatically reload/restart asterisk following IP change (dynamic IP)
Another useful script for those interested On the www.generationd.com web site you will now find the asteriskcontrol script file. This script can automatically restart Asterisk (gracefully) following a change in external IP address - for dynamic IP hosts. As well, it can update the SIP/IAX configuration files to reflect the new external IP addresses. You will have to edit the file config parameters for your settings... Enjoy!! -MD- ___ -- 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] Gold Mine CRM + Asterisk
We did a custom Goldmine customer lookup popup based on Asterisk CID...but that's about it. What are you trying to do? MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: March-24-08 9:05 AM To: Asterisk Users List Subject: [asterisk-users] Gold Mine CRM + Asterisk Hi, Has anyone used Gold Mine CRM with asterisk ? Thanks. Dovid ___ -- 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] Bug in voicemail's serveremail setting in 1.4.18
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in the voicemail messaging emailing operation. I had set serveremail option to: [EMAIL PROTECTED] and under ast 1.2.x messages arrived at user mailboxes from [EMAIL PROTECTED] . However, since upgrading emails arrive from [EMAIL PROTECTED] (the path from asterisk box to our corp mail server runs through firewall.mydomain.com). Before someone jumps to the obvious conclusion, I tried sending mail using sendmail [EMAIL PROTECTED] from the asterisk box, and the return address is correct!! So, Asterisk is somehow messing up the return address. It appears that as of asterisk 1.4.x, voicemail is incorrectly setting the return address - just how/why I don't know. The FROM name is properly set, and the from email is partially set right (voicemail@) but the remainder of the serveremail is wrong. Is there something else I need to change? Can someone explain this? Thanks, MD ___ -- 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] Bug in voicemail's serveremail setting in 1.4.18
For more info, I grab the relevant portion of the maillog. It looks like asterisk is trying to send using the right from email, but it's getting changed. This would suggest a sendmail problem, EXCEPT, it works fine when I send mail from the command line. Can anyone offer ideas? Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: [EMAIL PROTECTED], size=57772, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], [EMAIL PROTECTED] Mar 19 09:41:07 pbx sendmail[28459]: m2JDf7E1028459: from=[EMAIL PROTECTED], size=57908, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=pbx.MYDOMAIN.COM [127.0.0.1] Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: to=TEST testuser, [EMAIL PROTECTED] (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=87772, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (m2JDf7E1028459 Message accepted for delivery) Mar 19 09:41:07 pbx sendmail[28460]: STARTTLS=client, relay=[172.31.254.35], version=TLSv1/SSLv3, verify=FAIL, cipher=RC4-MD5, bits=128/128 Mar 19 09:41:07 pbx sendmail[28460]: m2JDf7E1028459: to=[EMAIL PROTECTED], delay=00:00:00, xdelay=00:00:00, mailer=smtp, pri=118089, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:41 AM To: Asterisk Users List Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in the voicemail messaging emailing operation. I had set serveremail option to: [EMAIL PROTECTED] and under ast 1.2.x messages arrived at user mailboxes from [EMAIL PROTECTED] . However, since upgrading emails arrive from [EMAIL PROTECTED] (the path from asterisk box to our corp mail server runs through firewall.mydomain.com). Before someone jumps to the obvious conclusion, I tried sending mail using sendmail [EMAIL PROTECTED] from the asterisk box, and the return address is correct!! So, Asterisk is somehow messing up the return address. It appears that as of asterisk 1.4.x, voicemail is incorrectly setting the return address - just how/why I don't know. The FROM name is properly set, and the from email is partially set right (voicemail@) but the remainder of the serveremail is wrong. Is there something else I need to change? Can someone explain this? Thanks, MD ___ -- 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] Bug in voicemail's serveremail setting in 1.4.18
Getting closerthis seems to be a sendmail issue not asterisk. It seems that if sendmail is run with -f (from) using any account other than root, the from domain is NOT trusted, and so sendmail does ns lookup - which of course resolves back to our firewall. So it seems that I need to tell sendmail to trust the asterisk account or voicemail from address From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:57 AM To: Asterisk Users List Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 For more info, I grab the relevant portion of the maillog. It looks like asterisk is trying to send using the right from email, but it's getting changed. This would suggest a sendmail problem, EXCEPT, it works fine when I send mail from the command line. Can anyone offer ideas? Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: [EMAIL PROTECTED], size=57772, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], [EMAIL PROTECTED] Mar 19 09:41:07 pbx sendmail[28459]: m2JDf7E1028459: from=[EMAIL PROTECTED], size=57908, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=pbx.MYDOMAIN.COM [127.0.0.1] Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: to=TEST testuser, [EMAIL PROTECTED] (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=87772, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (m2JDf7E1028459 Message accepted for delivery) Mar 19 09:41:07 pbx sendmail[28460]: STARTTLS=client, relay=[172.31.254.35], version=TLSv1/SSLv3, verify=FAIL, cipher=RC4-MD5, bits=128/128 Mar 19 09:41:07 pbx sendmail[28460]: m2JDf7E1028459: to=[EMAIL PROTECTED], delay=00:00:00, xdelay=00:00:00, mailer=smtp, pri=118089, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:41 AM To: Asterisk Users List Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in the voicemail messaging emailing operation. I had set serveremail option to: [EMAIL PROTECTED] and under ast 1.2.x messages arrived at user mailboxes from [EMAIL PROTECTED] . However, since upgrading emails arrive from [EMAIL PROTECTED] (the path from asterisk box to our corp mail server runs through firewall.mydomain.com). Before someone jumps to the obvious conclusion, I tried sending mail using sendmail [EMAIL PROTECTED] from the asterisk box, and the return address is correct!! So, Asterisk is somehow messing up the return address. It appears that as of asterisk 1.4.x, voicemail is incorrectly setting the return address - just how/why I don't know. The FROM name is properly set, and the from email is partially set right (voicemail@) but the remainder of the serveremail is wrong. Is there something else I need to change? Can someone explain this? Thanks, MD ___ -- 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] Bug in voicemail's serveremail setting in 1.4.18
RESOLVED! For others fiting a similar problem look at /etc/mail/service.switch This is the only way to force sendmail to not do a DNS lookup (first)... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:57 AM To: Asterisk Users List Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 For more info, I grab the relevant portion of the maillog. It looks like asterisk is trying to send using the right from email, but it's getting changed. This would suggest a sendmail problem, EXCEPT, it works fine when I send mail from the command line. Can anyone offer ideas? Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: [EMAIL PROTECTED], size=57772, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], [EMAIL PROTECTED] Mar 19 09:41:07 pbx sendmail[28459]: m2JDf7E1028459: from=[EMAIL PROTECTED], size=57908, class=0, nrcpts=1, msgid=[EMAIL PROTECTED], proto=ESMTP, daemon=MTA, relay=pbx.MYDOMAIN.COM [127.0.0.1] Mar 19 09:41:07 pbx sendmail[28458]: m2JDf7AJ028458: to=TEST testuser, [EMAIL PROTECTED] (0/0), delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=87772, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (m2JDf7E1028459 Message accepted for delivery) Mar 19 09:41:07 pbx sendmail[28460]: STARTTLS=client, relay=[172.31.254.35], version=TLSv1/SSLv3, verify=FAIL, cipher=RC4-MD5, bits=128/128 Mar 19 09:41:07 pbx sendmail[28460]: m2JDf7E1028459: to=[EMAIL PROTECTED], delay=00:00:00, xdelay=00:00:00, mailer=smtp, pri=118089, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: March-19-08 9:41 AM To: Asterisk Users List Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18 Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in the voicemail messaging emailing operation. I had set serveremail option to: [EMAIL PROTECTED] and under ast 1.2.x messages arrived at user mailboxes from [EMAIL PROTECTED] . However, since upgrading emails arrive from [EMAIL PROTECTED] (the path from asterisk box to our corp mail server runs through firewall.mydomain.com). Before someone jumps to the obvious conclusion, I tried sending mail using sendmail [EMAIL PROTECTED] from the asterisk box, and the return address is correct!! So, Asterisk is somehow messing up the return address. It appears that as of asterisk 1.4.x, voicemail is incorrectly setting the return address - just how/why I don't know. The FROM name is properly set, and the from email is partially set right (voicemail@) but the remainder of the serveremail is wrong. Is there something else I need to change? Can someone explain this? Thanks, MD ___ -- 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] VoIP Users Conference for Friday March 7th @ 12 Noon EST
I (like many others probably have) added the sender of the invite to my spam filter. That avoids the many replies - and also blocks future email from someone stupid enough to spam multiple entire list with an invite! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: March-06-08 5:39 AM To: Asterisk Users List Subject: Re: [asterisk-users] VoIP Users Conference for Friday March 7th @ 12 Noon EST Hi On Thu, Mar 06, 2008 at 09:29:01AM +0100, randulo wrote: Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT If you want to reply to this message regarding the schedule, please reply to the author. Your messages look very badly in the archives. And there is really no need to have 500 replies to this message on-list. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Aastra-Asterisk: 6 beeps then voice quality degrades
I have an unusual and recurring problem since I upgraded to Asterisk 1.4. Sometimes, mid-way through a call, I hear 6 shorts beeps and then the inbound voice quality degrades massively. It sounds like the other user is a robot...etc. I'm guessing something (aastra 480 or Asterisk 1.4) is warning me about a problembut what! Has anyone experienced this? MD ___ -- 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 and park/pickup feature
We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is setup like this: softkey4 type: park softkey4 label: Park softkey4 value: asterisk;70 softkey4 line: 1 softkey4 states: connected softkey4 type: pickup softkey4 label: Pickup softkey4 value: asterisk;70 softkey4 value: 1 softkey4 states: idle, outgoing (we also tried asterisk;700 with the same result). Has anyone got the softkey park/pickup working on aastra? Thanks Michelle ___ -- 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] Cisco 79xx users/consultants, 7970G color in particular share information
Are the 7921G phones convertable to SIP too? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sigma Networks Sent: March-01-08 11:56 AM To: Asterisk Users List Subject: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones composed of 7970, 7960 and 7906 phones. I've upgraded all of these to SIP 8-3-3SR2S and the basic functions are working. My current questions are: 1. How to remotely reboot 7970s. I have both web access and SSH access to the phones. The instructions I have for SSH are to use (1) user/pass (or whatever is in the confg) and then (2) debug/debug. Surprisingly reset is not a valid command to restart the phone. There doesn't appear to be a reset on the web page, maybe there's a hidden URL? 2. BusyLampField? Thanks in advance. ___ -- 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] Still can't pickup parked call
Well, I don't have a 701 extension defined but I do have _XXX which is where this call is jumping when I dial 701 to pickup. I have the include = parkedcalls above the _XXX definition, so I assumed that parked calls would be matched first. As well, since the 700 is matching to parked calls, I assumed 701 would as well. Still stuck -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore Sent: Tuesday, February 26, 2008 2:30 AM To: Asterisk Users List Subject: Re: [asterisk-users] Still can't pickup parked call I suspect there is something you are not telling us. Try posting this extension.conf file. Looking at the logs you have here leads me to believe you have an extension 701 defined to dial SIP/233. In other words, somewhere in your context is: exten = 701,1,Dial(SIP/233) or something very similar. An included context will never (ok, most probably won't ever) overwrite the definitions in the current context. For example, if you define extension 100 in your main context and then define extension 100 in an included context, the one in the main context will most probably always prevail. On Mon, Feb 25, 2008 at 9:41 PM, OCG Technical Support [EMAIL PROTECTED] wrote: I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have include = parkedcalls. The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and SIP/239-0915d5c8 -- Started music on hold, class 'default', on SIP/239-0915d5c8 == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 'SIPPeer/SIP/233-0915bf40ZOMBIE' in macro 'dialinternal' == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 'SIPPeer/SIP/233-0915bf40ZOMBIE' -- Started music on hold, class 'default', on SIP/239-0915d5c8 == Parked SIP/239-0915d5c8 on [EMAIL PROTECTED] Will timeout back to extension [entryocginternal] , 1 in 300 seconds -- SIP/233-0915bf40 Playing 'digits/7' (language 'en') -- SIP/233-0915bf40 Playing 'digits/0' (language 'en') -- SIP/233-0915bf40 Playing 'digits/1' (language 'en') -- Added extension '701' priority 1 to parkedcalls After parking the call, I then used that same phone to pickup 701 by dialing 701. As you can see, the 701 is being treated as a regular extension - not a parked call pickup. What is going on? Why is this nor working? -- Executing [EMAIL PROTECTED]:1] Macro(SIP/233-09152818, dialexternal|701|) in new stack ___ -- 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 -- Lacy Moore Somewhere I wish I wasn't ___ -- 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] Parked calls - can't pickup
It looks like I have a conflict! (See results of diaplan show below). How
can I force the parkedcalls context to be matched first? (I include
parkedcalls before I define the _X. priority).
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
'_X.' = 1. Macro(dialexternal|${EXTEN}|${dialaccount})
[pbx_config]
2. Goto(s|1)
[pbx_config]
[ Included context 'parkedcalls' created by 'res_features' ]
'701' = 1. ParkedCall(701)
[res_features]
-= 2 extensions (3 priorities) in 2 contexts. =-
pbx*CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jared Smith
Sent: Tuesday, February 26, 2008 9:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
I have 2 contexts in my extensions.conf: internal and
external calls.
I have included the parkedcalls context in both.
Do I need to preface the include with a # symbol?
No, you do not. You simply need a like that says:
include = parkedcalls
A couple of things to check:
1) make sure you haven't changed the context name in
features.conf (from parkedcalls to something else) and
2) you can always type dialplan show [EMAIL PROTECTED] from
the Asterisk CLI to see what would match if you dialed
extension 701 in that context.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Parked calls - can't pickup
Another clue...I repeated the dialplan show command for the 700 extension
and it too is listed AFTER the _X. match. However, forward a call to 700
works. Why would calling 701 not pickup the call? (Why is it matching the
_X. extension)
Thanks!
pbx*CLI dialplan show [EMAIL PROTECTED]
[ Context 'entryocginternal' created by 'pbx_config' ]
'_X.' = 1. Macro(dialexternal|${EXTEN}|${dialaccount})
[pbx_config]
2. Goto(s|1)
[pbx_config]
[ Included context 'parkedcalls' created by 'res_features' ]
'700' = 1. Park()
[res_features]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
OCG Technical Support
Sent: Tuesday, February 26, 2008 10:02 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
It looks like I have a conflict! (See results of diaplan
show below). How can I force the parkedcalls context to be
matched first? (I include parkedcalls before I define the
_X. priority).
pbx*CLI dialplan show [EMAIL PROTECTED] [ Context
'entryocginternal' created by 'pbx_config' ]
'_X.' = 1. Macro(dialexternal|${EXTEN}|${dialaccount})
[pbx_config]
2. Goto(s|1)
[pbx_config]
[ Included context 'parkedcalls' created by 'res_features' ]
'701' = 1. ParkedCall(701)
[res_features]
-= 2 extensions (3 priorities) in 2 contexts. =- pbx*CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: Tuesday, February 26, 2008 9:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
On Mon, 2008-02-25 at 21:03 -0500, Michelle Dupuis wrote:
I have 2 contexts in my extensions.conf: internal and
external calls.
I have included the parkedcalls context in both.
Do I need to preface the include with a # symbol?
No, you do not. You simply need a like that says:
include = parkedcalls
A couple of things to check:
1) make sure you haven't changed the context name in features.conf
(from parkedcalls to something else) and
2) you can always type dialplan show [EMAIL PROTECTED] from the
Asterisk CLI to see what would match if you dialed extension 701 in
that context.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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[asterisk-users] Still can't pickup parked call
I'm still struggling to pickup calls. I now have a single context (entryocginternal) where I have include = parkedcalls. The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and SIP/239-0915d5c8 -- Started music on hold, class 'default', on SIP/239-0915d5c8 == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 'SIPPeer/SIP/233-0915bf40ZOMBIE' in macro 'dialinternal' == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 'SIPPeer/SIP/233-0915bf40ZOMBIE' -- Started music on hold, class 'default', on SIP/239-0915d5c8 == Parked SIP/239-0915d5c8 on [EMAIL PROTECTED] Will timeout back to extension [entryocginternal] , 1 in 300 seconds -- SIP/233-0915bf40 Playing 'digits/7' (language 'en') -- SIP/233-0915bf40 Playing 'digits/0' (language 'en') -- SIP/233-0915bf40 Playing 'digits/1' (language 'en') -- Added extension '701' priority 1 to parkedcalls After parking the call, I then used that same phone to pickup 701 by dialing 701. As you can see, the 701 is being treated as a regular extension - not a parked call pickup. What is going on? Why is this nor working? -- Executing [EMAIL PROTECTED]:1] Macro(SIP/233-09152818, dialexternal|701|) in new stack ___ -- 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
