Re: [asterisk-users] SLA (Shared Line Appearance) and realtime
Leandro Dardini gmail.com> writes: > > Hello,do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? > > Leandro > > Are there any updates here? This would be a great to implement. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need Help
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business's phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over the Holidays and as of right now that's about all we have working right now are the phones. The reason I joined this list is because I was hoping to get our external paging intercom system back up and running (it runs off of a sound card but cant get it all configured correctly) and to be honest I have no clue where to start. I've tried reading some online guides but nothing. [MRKlogoblkemail] Joe Ruffolo Director of Operations 801 N State St Unit C Elgin, Il. 60123 847-468-1700v 847-468-0717f j...@mrkgroup.com<mailto:j...@mrkgroup.com> www.mrkgroupltd.com<http://www.mrkgroupltd.com/> [mrk r2 logo] <><>-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
> On Mon, 28 Mar 2011 08:20:23 -0400, vip killa > wrote: > >Is anyone using asterisk with fail2ban? > > Sorry for hi-jacking the thread, but I was wondering if there were a > lighter alternative that I could run on appliances? > > Python uses too much RAM, but I need to find a way to ban hackers from > trying to connect to Asterisk from the Net. I had worked with the sshguard guys to add support for Asterisk; I believe they added basic support. I haven't gotten around to revisiting that issue just yet so I don't know for sure. http://lists.digium.com/pipermail/asterisk-users/2010-December/256928.html sshguard is *extremely* lightweight compared to most things; it's a very efficient compiled C application that doesn't have (m?)any dependencies. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
I am as well - Original Message - From: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tue Feb 01 11:22:41 2011 Subject: Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3? Hi again, Nobody knows how to disable it? Can at least someone pinpoint me where can I check the latest documentation regarding SRTP. Maybe something might have change in the meanwhile 'Cause so far it looks like there is a bug in asterisk. Well, maybe I should report this bug then. - Miguel Baptista On 28.01.2011 18:22, Miguel Baptista wrote: Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the encryption=no in sip.conf and the _SIPSRTP_CRYPTO=disable on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have noload=res_srtp.so on my modules.conf otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP attacks and sshguard
> On Thu, Dec 09, 2010 at 07:57:37AM -0600, Joe Greco wrote: > > Specifically looking for examples of (or how to generate) > > > > 1) .*No registration for peer '.*' (from ) > > 2) .*Host failed MD5 authentication for '.*' (.*) > > 3) .*Failed to authenticate user .*@.* > > > > If anyone who is more familiar with the attacks or how to generate > > these messages would give me some assistance, or chime in on the > > sshguard-users list, that'd be most appreciated. > > You could use SIPVicious to run attacks on your own servers: > http://code.google.com/p/sipvicious/ Those tools don't seem to generate (or I can't figure out how to get them to generate) any of the above messages; I already have plenty of the Registration from 'foo' failed for 'host' - reason messages that sipvicious seems to generate. I'm not quite sure what to do to generate examples of the above messages, any suggestions are appreciated. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP attacks and sshguard
Hello, We had been seeing SIP-guessing attacks on our Asterisk server here. While it wasn't that hard to write a once-a-minute cron job to spank the lusers, that runs once a minute and creates little spikes in the usage and I/O graphs, and is slower to respond than I'd really prefer. I felt that it'd be much cooler to get something more comprehensive put together. We don't use fail2ban because I don't like having to install python. sshguard is a high-performance compiled C application that can run off a log file or a pipe from syslogd to sshguard, meaning that it can respond a lot more quickly than once a minute, and works with very modest overhead on the host system. It also has features such as touchiness, so that it can get tougher on a miscreant as time goes on; my own shell script is naive in that once it passes a threshold, there's just a permanent rule generated. This worries me if I ever have a situation where a legitimate remote client gets messed up and tries the wrong password or something like that; sshguard does a much nicer job in this regard. In any case, my initial attempts to create rules for sshguard didn't work right, quite possibly because I don't often work in LEX/YACC. I submitted a request to the sshguard guys suggesting new rules. http://www.sshguard.net/support/submission/detail/49ce7182028d8b6f3e3d/ and on their mailing list, a little more: http://sourceforge.net/mailarchive/forum.php?thread_name=F4E10075-5D93-43B4-B73A-1FD217BE130D%40sshguard.net&forum_name=sshguard-users In particular, they're looking for log examples of some of those messages, but I have no idea how to generate the conditions that would cause these messages. I'm also not sure if there's a way to disable color codes in the Asterisk log files; we log indirectly via BSD's "logger" # asterisk -vvv 2>&1 | logger -t asterisk so it may be thinking that the console is color-capable. We use this method because this forces them through the syslog mechanism; we need that for centralized logging, and it's handy for things like sshguard too. Specifically looking for examples of (or how to generate) 1) .*No registration for peer '.*' (from ) 2) .*Host failed MD5 authentication for '.*' (.*) 3) .*Failed to authenticate user .*@.* If anyone who is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. Thanks. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Contacts via Asterisk?
I've done a remote door unlock system in the past. The customer had an existing magnetic lock system that utilized push buttons on the wall to release the magnetic locks on the doors. They already had this system, and the associated door controller. I used an APC AP9210 Master Switch network power controller to solve the problem. I plugged the power brick for the door controller into the AP9210, put the AP9210 on the network, and wrote a script that caused the device to cut power to the door controller for 5 seconds by setting an SNMP variable. I coupled that shells script with some dialplan logic that asked the caller for a pin, then using the callerid(num) value as a username, checked the username/pin combo against a database.. If the combination passed, the shell script was called. If it didn't, I looped back to the request for the pin, for up to 2 more times, then disconnected the call. I know Cyberdata has some SIP based door intercoms that have lock relays built in, as well. I have no idea what the cost for them is, though. On 11/15/2010 1:35 PM, Cassius Smith wrote: > Hi all, > I have had (what I consider) an odd request. The installation I'm > working on now is an office on a multi-floor building. They 're looking > for some kind of solution with the phone system to provide door control. > We are a non-profit so of course I'm looking for something VERY inexpensive. > > I'm sure /someone/ has done something like this. I'd appreciate any ideas. > > Cassius Smith > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP 800 Origination/Termination - International
Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't handle Canadian inbound calls to that number, so I need to find another provider that can. Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
-- Forwarded message --
From: Joe Wood
Date: Thu, Aug 26, 2010 at 6:58 PM
Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw
file using BackGround() but no audio is heard from the phone
To: Asterisk Users Mailing List - Non-Commercial Discussion
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload => format_mp3.so
preload => codec_ulaw.so
preload => format_pcm.so
My extensions.conf looks like:
[general]
autofallthrough=yes
static=no
writeprotect=no
extenpatternmatchnew=yes
clearglobalvars=no
[conference-calls]
exten => s,1,Answer()
exten => s,n,Background(welcome)
exten => s,n,Background(and)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,Background(conference-reservations)
exten => s,n,Wait(2)
exten => s,n,Background(enter-conf-pin-number)
exten => s,n,WaitExten(10)
exten => i,1,Playback(pbx-invalid)
exten => i,n,Goto(conference-calls,9000,1)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => ${EXTEN},1,Meetme(${EXTEN})
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer("SIP/2063161626-0001", "") in new stack
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer("SIP/2063161626-0002", "") in new stack
-- Executing [...@conference-calls:2]
BackGround("SIP/2063161626-0001", "welcome") in new stack
-- Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
BackGround("SIP/2063161626-0002", "welcome") in new stack
-- Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround("SIP/2063161626-0001", "and") in new stack
-- Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround("SIP/2063161626-0002", "and") in new stack
-- Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:4]
BackGround("SIP/2063161626-0001", "thank-you-for-calling") in new
stack
-- Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:4]
BackGround("SIP/2063161626-0002", "thank-you-for-calling") in new
stack
-- Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:5]
BackGround("SIP/2063161626-0001", "conference-reservations") in
new stack
-- Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:5]
BackGround("SIP/2063161626-0002", "conference-reservations") in
new stack
-- Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:6]
Wait("SIP/2063161626-0001", "2") in new stack
-- Executing [...@conference-calls:6]
Wait("SIP/2063161626-0002", "2") in new stack
-- Executing [...@conference-calls:7]
BackGround("SIP/2063161626-0001", "enter-conf-pin-number") in new
stack
-- Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:7]
BackGround("SIP/2063161626-0002", "enter-conf-pin-number") in new
stack
-- Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:8]
WaitExten("SIP/2063161626-0001", "10") in new stack
-- Executing [...@conference-calls:8]
WaitExten("SIP/2063161626-0002", "10") in new stack
-- Timeout on SIP/2063161626-0001, going to 't'
-- Executing [...@conference-calls:1]
Playback("SIP/2063161626-0001", "vm-goodbye") in new stack
-- Playing 'vm-goodbye.ulaw' (language 'en')
-- Timeout on SIP/2063161626-0002, going to 't'
-- Executing [...@conference-calls:1]
Playback("SIP/2063161626-0002", "vm-goodbye") in new stack
-- Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
Hangup("SIP/2063161626-0001", "") in new stack
== Spawn extension (conference-calls, t, 2) exited non-zero on
'SIP
[asterisk-users] Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload => format_mp3.so
preload => codec_ulaw.so
preload => format_pcm.so
My extensions.conf looks like:
[general]
autofallthrough=yes
static=no
writeprotect=no
extenpatternmatchnew=yes
clearglobalvars=no
[conference-calls]
exten => s,1,Answer()
exten => s,n,Background(welcome)
exten => s,n,Background(and)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,Background(conference-reservations)
exten => s,n,Wait(2)
exten => s,n,Background(enter-conf-pin-number)
exten => s,n,WaitExten(10)
exten => i,1,Playback(pbx-invalid)
exten => i,n,Goto(conference-calls,9000,1)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => ${EXTEN},1,Meetme(${EXTEN})
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer("SIP/2063161626-0001", "") in new stack
== Using SIP RTP CoS mark 5
-- Executing [...@conference-calls:1]
Answer("SIP/2063161626-0002", "") in new stack
-- Executing [...@conference-calls:2]
BackGround("SIP/2063161626-0001", "welcome") in new stack
-- Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
BackGround("SIP/2063161626-0002", "welcome") in new stack
-- Playing 'welcome.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround("SIP/2063161626-0001", "and") in new stack
-- Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:3]
BackGround("SIP/2063161626-0002", "and") in new stack
-- Playing 'and.ulaw' (language 'en')
-- Executing [...@conference-calls:4]
BackGround("SIP/2063161626-0001", "thank-you-for-calling") in new
stack
-- Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:4]
BackGround("SIP/2063161626-0002", "thank-you-for-calling") in new
stack
-- Playing 'thank-you-for-calling.ulaw'
(language 'en')
-- Executing [...@conference-calls:5]
BackGround("SIP/2063161626-0001", "conference-reservations") in
new stack
-- Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:5]
BackGround("SIP/2063161626-0002", "conference-reservations") in
new stack
-- Playing
'conference-reservations.ulaw' (language 'en')
-- Executing [...@conference-calls:6]
Wait("SIP/2063161626-0001", "2") in new stack
-- Executing [...@conference-calls:6]
Wait("SIP/2063161626-0002", "2") in new stack
-- Executing [...@conference-calls:7]
BackGround("SIP/2063161626-0001", "enter-conf-pin-number") in new
stack
-- Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:7]
BackGround("SIP/2063161626-0002", "enter-conf-pin-number") in new
stack
-- Playing 'enter-conf-pin-number.ulaw'
(language 'en')
-- Executing [...@conference-calls:8]
WaitExten("SIP/2063161626-0001", "10") in new stack
-- Executing [...@conference-calls:8]
WaitExten("SIP/2063161626-0002", "10") in new stack
-- Timeout on SIP/2063161626-0001, going to 't'
-- Executing [...@conference-calls:1]
Playback("SIP/2063161626-0001", "vm-goodbye") in new stack
-- Playing 'vm-goodbye.ulaw' (language 'en')
-- Timeout on SIP/2063161626-0002, going to 't'
-- Executing [...@conference-calls:1]
Playback("SIP/2063161626-0002", "vm-goodbye") in new stack
-- Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@conference-calls:2]
Hangup("SIP/2063161626-0001", "") in new stack
== Spawn extension (conference-calls, t, 2) exited non-zero on
'SIP/2063161626-0001'
-- Executing [...@conference-calls:2]
Hangup("SIP/2063161626-0002", "") in new stack
== Spawn extension (conference-calls, t, 2) exited non-zero on
'SIP/2063161626-0002'
Has anyone else encountered this problem before? I saw one posting on
the listserv, but it said to add in the pcm lib and I did that and no
change.
Help.
Thanks a bunch,
Joe
--
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Re: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks
sg01*CLI> core show version Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-16 15:17:26 UTC On Wed, Aug 18, 2010 at 10:19 AM, Danny Nicholas wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Wood > Subject: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for > allinbound trunks > > > > Since you can see the CLI log, please post your asterisk version (core show > version) so we can see what flavor of Asterisk your "AN" is operating under. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks
Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe -- Forwarded message -- From: Joe Wood Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisk...@lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering s@ as an extension? 18 Aug 00:01:09.301/GLOBAL/ser: RECEIVED message from 209.221.186.51:5060: REGISTER sip:209.221.186.98 SIP/2.0 Via: SIP/2.0/UDP 209.221.186.51:5060;branch=z9hG4bK41fb6b8f;rport Max-Forwards: 70 From: ;tag=as7608 To: Call-ID: 5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.6.2.11 Authorization: Digest username="2063161626", realm="pugetsoundtelecom.net", algorithm=MD5, uri="sip:209.221.186.98", nonce="4c6b8544f41a0643a65f4e17199268a018b32070", response="dcbb7adca43c6c7455c9942010c84423", qop=auth, cnonce="45cc3d6b", nc=0002 Expires: 120 Contact: Content-Length: 0 18 Aug 00:01:09.301/5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50/ser: processing REGISTER received from 209.221.186.51:5060 18 Aug 00:01:09.302/5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50/ser: saving contact sip:s...@209.221.186.51 into the database 18 Aug 00:01:09.302/GLOBAL/ser: SENDING message to 209.221.186.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.221.186.51:5060;branch=z9hG4bK41fb6b8f;rport=5060 From: ;tag=as7608 To: ;tag=5fceb36a80dbc27aca680924f1b8b505-19ad Call-ID: 5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50 CSeq: 104 REGISTER Contact: ;expires=300 Server: Sippy Softswitch v2.0.80 Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wrote: > On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood wrote: >> >> I don't see any >> >> On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby >> wrote: >> > >> > You don't have any extensions in your default context that match the >> > extension that your sip peer is dialing in on. 's' is not a default >> > extension for SIP...try using _X., and see what you get. Bump up the >> > CLI >> > (core set verbose 10) and then repost a failed called attempt. Some SIP >> > providers also use a + symbol in front of their inbound calls, so you >> > may >> > need to use _+X., instead. >> >> I don't see any call attempt/logs when I bump up the verbosity, and >> when I check my verbose logs I show: >> > > The next step would be to enable sip debug on the peer you're trying to > receive calls from (sip set debug peer PEERNAME or sip set debug ip > IPADDRESS). Then send another call inbound and see what's happening. As > far as the 's' extension, that's the "start" extension, it's used when no > other extension information is presented. Pretty much every SIP peer I've > ever seen presents an extension when entering a context, and thus the 's' > extension doesn't catch it. I've typically only seen 's' used in Macros and > with inbound analog lines. > My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: <--- SIP read from UDP:209.221.186.98:5060 ---> INVITE sip:s...@209.221.186.50 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 Max-Forwards: 16 From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 To: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy Softswitch v2.0.80 cisco-GUID: 1225641884-3786690633-966044271-4144140181 h323-conf-id: 1225641884-3786690633-966044271-4144140181 Content-Length: 321 Content-Type: application/sdp v=0 o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 s=- c=IN IP4 209.221.186.98 t=0 0 m=audio 60304 RTP/AVP 0 a=fmtp:4 bitrate=6300;annexa=no a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000 a=oldmediaip:208.76.155.20 a=nortpproxy:yes <-> [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 1 [ 75]: Record-Route: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 2 [ 85]: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 3 [ 94]: Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 4 [ 16]: Max-Forwards: 16 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 5 [ 85]: From: 2538544199 ;tag=f7093e2d7e16a927d0816f6f5ed7aba4 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 6 [ 35]: To: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 8 [ 16]: CSeq: 200 INVITE [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 9 [ 55]: Contact: Anonymous [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 10 [ 12]: Expires: 300 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 11 [ 36]: User-Agent: Sippy Softswitch v2.0.80 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 14 [ 19]: Content-Length: 321 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 15 [ 29]: Content-Type: application/sdp [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 16 [ 0]: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 0 [ 3]: v=0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 2 [ 3]
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wrote: > > You don't have any extensions in your default context that match the > extension that your sip peer is dialing in on. 's' is not a default > extension for SIP...try using _X., and see what you get. Bump up the CLI > (core set verbose 10) and then repost a failed called attempt. Some SIP > providers also use a + symbol in front of their inbound calls, so you may > need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'default' (0xb77980c0) in local table 0xb77960c0; registrar: pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 1 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 2 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 3 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 4 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 5 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 6 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 7 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 8 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0; registrar: features [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700' priority 1 to parkedcalls (0xb7797ee0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.89 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints and swap in new dialplan: 0.02 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old dialplan: 0.11 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Total time merge_contexts_delete: 0.000102 sec [Aug 4 19:17:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:19:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:21:39] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 I get the same error. Same random voicemail when no voicemail is configured. I was under the impressing that "s" was the catchall for all incoming trunks. What has changed? Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =>1. Wait(1)[pbx_config] 2. Answer() [pbx_config] 3. Background(welcome)[pbx_config] 4. Background(and)[pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations)[pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: "Extension is unavailable. Please leave your message after the tone". sip.conf: [general] register => NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten => s,1,Wait(1) exten => s,n,Answer() exten => s,n,Background(welcome) exten => s,n,Background(and) exten => s,n,Background(thank-you-for-calling) exten => s,n,Background(conference-reservations) exten => s,n,Waitfor() exten => s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SwitchVox AA355 w/ 4 Port PRI and 2 Port FXO and 2 Port FXS For Sale on eBay
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=230492577678#ht_500wt_1076 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big time system
Cary- Asterisk may carry you a way down this road, but in the end, it's not, and was never designed to be a class 5 telecom switch. There are people working on a carrier grade implementation that may or may not be fully class 5, but I don't know what the status is on that. I haven't gotten an answer from Digium on that lately. What you're looking for are local gateways that backhaul to a central switch site with equipment that can support traffic from multiple rate centers in multiple LATAs. This gets complicated quickly, especially if your rate centers are spread across multiple states. You'll want some type of Multiservice Access Platform (MSAP). Zhone makes the MALC and their newer MXK box. Adtran has the TA-5000 shelf. Neither are what you'd call cheap. Both will provide T1 access, DSL, SDSL, VDSL, bonded, and even ethernet access to the customer over a variety of transport options, including copper pairs. The Zhone box already has SIP backhaul for voice traffic, and the Adtran shelf should have it soon. Today the Adtran box has GR303 backhaul for voice. All that said, what you're proposing indicates to me that you're likely to need to establish CLEC certification in whatever states you'll be operating. That in itself is not a short process. It can take anywhere from 90 days to a year depending on the state, and expect to spend from $10K up on legal costs per state alone. Insurance, financial health, and other requirements vary by state as well. The ILECs generally won't even talk to you about establishing colo and gaining access to the copper loops until you get the CLEC certificate. Generally the process starts by getting the certificate, then negotiating an ICA, then trunking services, then colo. Different carriers will be easier to work with than others, but they are all a pain. AT&T requires you to have a $10M general liability policy in place before you can even submit a request for a space availability report. All this is not to say it can't be done, but to point out that it's a very difficult process to negotiate, even when you have done it several times. Without experience it can be close to impossible. I'd suggest getting a good telecom/clec consultant and a good telecom lawyer (I know a few) involved early in the process, or you'll end up spending ALOT of money. Hit me off-list and I can give you more info. Joe On 6/24/2010 11:24 PM, Cary Fitch wrote: > We are an asterisk user... small time system 50-100 users or so. > > But, we have an opportunity to get into a big time telecom activity. > > It would have 2000 to 30,000 user lines per city, and we would like to have > those brought back to a central location for control and because transport > can be more economical than remote site rentals, maintenance and personnel. > > We could take the local lines into concentrators (TNTs or equivalent) and > bring back IP to a central site, or put servers at the remote cities. > > Our object is to serve as a "central office switch" for subscribers on > standard telco service loops. > > This isn't a "How many lines can I handle using a Belchfire 2600 processor?" > type question but a request for pointers to big time systems. There would > be no IP path to the end user, "just" copper. > > Thank you > Cary Fitch > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)
The simple fix your missing is to simply put your NICs on different layer 3 segments. In a configuration wherein multiple adapters are bound to the same layer 3 network (subnet), most IP stacks will only send traffic out the first NIC to bind to that network. I've seen this many times in data centers where server guys configure the NICs in the same subnet for "load balancing". With some IP stacks, even arp responses are only sent from the first NIC to bind to the subnet. This creates problems at layer 2, especially in large traffic volume situations. Since a technically correct arp response is sent - the mac address in the response is for the NIC to which the arp request was sent - from the first NIC to bind to the subnet, the switch(es) never actually learn where the MAC addresses for the other NICs are located. This creates a situation where all traffic for these NICs is broadcast to all ports in the VLAN. Joe On 6/1/2010 10:27 AM, Mike wrote: > Hi, > > Reward offered: 50$ (paypal), and I am sure this is a ridiculous thing I > have missing. > > My goal: On a 2 NIC Asterisk box, to send packets that came in Asterisk > on NIC1 back to NIC 1, and NIC2 back to NIC 2. (basically, send them > back the same way they came from). > > I have been doing what was recommended to me and mangling packets left > and right. I have reached a point where I am stuck, and can`t imagine > why this last little step isnt working. > > As you know, Asterisk sends all packets "from" the default IP (in my > case, NIC 1 IP). So connections to NIC 1 work fine, to NIC 2 they don`t. > I therefore put in some routing rules to help me. Some example, a phone > (remote PBX setup) coming in from 65.77.77.77 works fine because of > these (slightly obfuscated by changing IPs shown) routing rules: > > ip rule show: > > 0: from all lookup 255 > > 32759: from all fwmark 0x64 lookup ISP2 (<- this is key to my issue) > > 32760: from all to 65.77.77.77 lookup ISP2 > > 32766: from all lookup main > > 32767: from all lookup default > > ip route show table ISP2: > > default via 22.22.22.22 dev eth1 src 22.22.22.21 > > BUT I can't reliably know where the phones come from (long story), or > what IP they use (ISP1 or ISP2) to connect to me. So instead, I have > done this with iptables: > > MARK all -- anywhere STRING match "22.22.22.21" ALGO name bm TO > 65535MARK set 0x64 > > Basically saying to mark all packets that have the string "22.22.22.21" > in it's SIP content (meaning they came in on NIC2 originally because the > phone registered to 22.22.22.21) with mark 0x64. And that works fine, > because another rule that LOGs these marked packets is logging them > correctly. > > Because of my above routing rules, packets going out marked with 0x64 or > those going to 65.77.77.77 should go to the same ip route (route table > ISP2). Mysteriously, I see that packets going to 65.77.77.77 (using > wireshark) are correctly mangled as coming from 22.22.22.21, but not > those marked with 0x64. Those still go out through the default routing > table. > > What the heck am I missing? I believe I have done my homework, but there > is no more door left to bang my head on. > > Mike > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - providers discontinuing support
> What is wrong with IAX2 protocol? > If IAX2 is so much better than SIP so why providers discontinuing support for > IAX2 > > I was with provider "callwithus" but they discontinue IAX2 > I switched to "checkbox.cc" but they discontinued it as well. > > What is wrong with IAX2? The same thing that's wrong with a lot of theoretically superior technologies: SIP is *more* universal, and therefore if it's a choice of supporting two technologies or just one, SIP has more bang for the buck. Almost every gadget or gizmo supports SIP. Few support IAX2. To support IAX2 for the relatively small number of people who know what it is and who are running Asterisk or IAX2-capable gear may be more trouble than it is worth. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res fax help
I have res_fax setup and working for the most part. However, I'm seeing some fax machines drop the connection on me - Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel 'DAHDI/1-1' did not return a frame; probably hung up. -- Channel 0/1, span 1 got hangup, cause 102 -- Channel 'DAHDI/1-1' FAX session '20' is complete, result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x98', transfer rate: '14400', remoteSID: '' == Spawn extension (macro-fax_rcv, s, 3) exited non-zero on 'DAHDI/1-1' in macro 'fax_rcv' It appears to be dropping out of my macro fax_rcv at that point and not executing the next step in the dialplan, which is a System call to a script that converts the tif to a pdf and emails it to the extension owner. My question is how do I ensure that my script is called when the far end hangs up before the call progresses that far in the dialplan? My first thought is to add something like this- exten => h, 1, System(callmyscript.sh,arg1,arg2,arg3,arg4,argimapirate) to the macro, but I'm not sure if that would do it or not. Anyone have any thoughts? Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent Callback methods?
Since AgentCallbackLogin() was apparently removed from 1.6, does anyone have anything to replace that functionality? Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
> Hi All, > > Anyone one info of where I can get a 'free' DID number ? > > I have setup my asterisk box (home) and want to learn more but I need a #. I highly suggest http://tinyurl.com/ya9vzsa ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good dialplan code out there to implementvertical service codes?
At the moment, 1.6.0.20 realtime with Dahdi 2.2, TDM is a TE420, but that won't be customer facing. Thanks- Joe Danny Nicholas wrote: > It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are > using. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman > Sent: Monday, December 28, 2009 8:14 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Any good dialplan code out there to > implementvertical service codes? > > Greetings- > > I'm in the process of turning up an Asterisk box for a customer and was > wondering if anyone had any good code they could share for implementing > vertical service codes within Asterisk. I'd really rather not have to > spend hours making new wheels if someone has one or more that will fit. > > One of my issues is that I've had a very hard time finding out exactly > which VSC's are already implemented in what parts of Asterisk. This > system will be for the most part an IP Centrex platform for this > customer, who will be selling services to his end-users. > > I've got a pretty good line on how I'm going to handle personal LD pin > codes using ODBC, but if anyone has any code for that I might get some > pointers from, I'd appreciate that too. > > Any thoughts or suggestions are greatly appreciated. > > > Thanks- > Joe > > ___ > -- 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] Looking at Asterisk for 8000sq/ft residential
> Rick Huebner wrote: > > My brother-in-law is finishing up his McMansion and I've done all of the > > low voltage wiring and am starting the trimout. We are batting around > > what to do for a phone system and I'm torn between a Panasonic > > TAW824/TVA50 and using an Asterisk implementation. I'm very strong on > > the networking/linux/basic hacking(old school, not criminal) side. I've > > downloaded the Asterisk VM and have some implentation questions before > > we make a decision. Of course we are running out of time because I need > > to order either RJ-11 or RJ-45 keystones for the plates to finish the > > trim out. > > You can use the 8 position modular jacks regardless ( misnamed RJ-45 ) > so that should not stop you from finishing the trim out. Replacing the jacks down the road is (yawn). Big deal if you need to. Pick whatever works for the current deployment. You WILL be running at least Cat5e to the jacks, every jack, right? That's the problematic bit for the future - not the jack itself. You don't want to plug RJ11 into RJ45 jacks, so plan to make it work out somehow. You can always make RJ11/45 cables so RJ45 is my preference. > The Panasonic systems I have used over the last 20 years are rugged, > hang on the wall, connect with proper protection and forget them for > years on end. They all have had dual ports that will either use a POTS > single line phone, or one of their multibutton phones without any > rewiring, reprogramming, and many even support one of each per port. > An ideal system for a large house. I will second that the Panasonic systems are nice. They're everything you would expect in a proprietary phone system and you are not likely to be disappointed in the system as long as it offers the features you want. That said, you are forever locked into that system. I can't say I've looked much into the Panasonics since we went Asterisk here, but at the time there was a definite feeling of it being "last decade's" technology, and that was the better part of a decade ago. Features will "just work." But the features you don't have, you will never have, at least not without a lot of hacking. Asterisk is forward-looking. Expect things not to work without some programming and configuration. Consider something simple like VoIP. With Asterisk, easy to use VoIP on either or both sides. I wouldn't have guessed years ago that one day my cell phone would double as a SIP extension. Yet it happened and it works. And look at where telephony is going. It's going to be VoIP, sooner or later. All that copper's going away. Get network-centric now and maybe you won't get stuck with a dinosaur. It's a tradeoff. Asterisk can be more work. But you can do more too. Our Asterisk announces calls by name over the intercom and lights it up on the TV's as well. Easy to do with a programmable box. > Although many will disagree, for most users Panasonic systems with > normal requirements work well for long periods with no problems and > have lots of features. > For the geek who wants to play, drive the rest of the family nuts > changing things, then consider Asterisk. There's nothing saying you have to change things constantly with Asterisk. Some of us have systems that are generally untouched for long periods of time. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] Any good dialplan code out there to implement vertical service codes?
Greetings- I'm in the process of turning up an Asterisk box for a customer and was wondering if anyone had any good code they could share for implementing vertical service codes within Asterisk. I'd really rather not have to spend hours making new wheels if someone has one or more that will fit. One of my issues is that I've had a very hard time finding out exactly which VSC's are already implemented in what parts of Asterisk. This system will be for the most part an IP Centrex platform for this customer, who will be selling services to his end-users. I've got a pretty good line on how I'm going to handle personal LD pin codes using ODBC, but if anyone has any code for that I might get some pointers from, I'd appreciate that too. Any thoughts or suggestions are greatly appreciated. Thanks- Joe ___ -- 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] FW: hi Dan
> What say you to the proposal that some approaches to seeking help are > so ridiculous they should not be tolerated? > > Community standards neither conceive nor enforce themselves. This community standard is entirely self-enforcing. If everybody thinks the request for help is unwarranted and doesn't deserve an answer, then nobody answers. If it isn't so intolerable to fall victim to that, then someone may feel inclined to help out and answer. If the volume of such requests becomes sufficiently burdensome that it exhausts those answering the questions, then eventually the equilibrium resets; those answering the questions begin to pick out the ones that they feel are worthy of answers. Years ago, I got a bit of a panicky phone call from a sysadmin at an ISP that I was loosely familiar with from mailing lists. He was rather frazzled and puzzled because he had been struggling to solve a problem during a downtime; it was something I was familiar with and had been advocating on a mailing list. He was doing something completely reasonable-seeming, it's just that what he was doing didn't work, and had never worked that way. I walked him through a different method, from memory, solved his problem. I'm positive I could have charged him billable hours, but I didn't, because I felt somewhat responsible for having advocated something rather complex that was followed by a competent admin and it blew up in his face - precisely *because* the problem and fix were obscure. The Asterisk community is great at promoting itself, but quite frankly the documentation and solutions are sometimes not all that great. It can be challenging to find the right fix, or even *a* fix. Questions *must* be expected. The community has generally been fairly successful at coping with the questions; I view the beginning of this thread to be a sign of that. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] FW: hi Dan
> Sorry, I can't resist. > > How do I join the Mail List Nazi Corp? Do I have to be invited, or can I > just self appoint myself? Asking neophyte questions are objected to by > some, top posting by those who blast others, etc. > > How about leaving member chastisement to the sponsor of the list? That's unlikely to happen in most cases. > Some people have no one within 250 miles of where they are to learn from or > learn better by working with code than reading inscrutable examples from > different versions, and other inanimate pages of examples that have "wrong" > variables, etc. Yes. > Nearly everyone can be criticized for something, Asking "dumb" questions, > top posting, bottom posting and leaving 3 pages of "crap" to scroll through, > answering questions that were answered 5 posts down, because they didn't > review the newer messages before posting, and more. > > Be charitable and kind. Have a nice day. There's absolutely something to be said for that. On the other hand, there is also something to be said for making people exhaust the available resources prior to solving their problems for them. You can even be charitable and kind while doing so... Back in the '90's, I knew a really bright guy who knew Windows and Novell inside and out. He was just learning UNIXy stuff (FreeBSD in particular) and he was discovering that there was a lot of application for the stuff. He would frequently approach me, desperately seeking an answer to some general problem of some sort. I would typically give relatively vague answers, ending up essentially with a "figure it out yourself." This frustrated him to no end, but he would do so. Later, he would come to me, almost always with a workable solution, at which point we would often discuss the ins and outs of several different options. His solution wasn't always the *best*, but it would always serve as a foundation for the rest. Years later, he thanked me. At the time, he didn't really appreciate what I was doing and didn't see the bigger picture. Looking back on it, I think he saw that I had always tried to aim him in a sensible direction before shoving him off on his own to figure it out. He eventually grew confident enough and capable enough that he would no longer need to ask for help. I can fix your problems for you, or I can teach you to be self- sufficient... which one is doing you more of a favor? It may seem more "charitable and kind" to simply give someone answers, but I do not think it actually is, at least in this sort of situation. As for the original poster? It's my impression, reading in between the lines, that he probably hasn't tried that hard. Asterisk on Linux is pretty straightforward, and MOH is probably not that rough to get running. On FreeBSD? That's a different thing. Bleh. But it's still better to do it on-list rather than selecting someone at random to go and bother. I don't think anyone will prevent you from being "charitable and kind" by providing answers to the guy's questions on the list though. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
> By "fast" I mean the best Business DSL Bellsouth has to offer: "Up to > 6.0 Mbps downstream - Up to 512 Kbps upstream" That almost sounds like an invitation to check out what business service your cableco offers. One thing to be aware of with DSL and cable modems is that there can be various ill effects as your line gets closer to its rated capacity; do not expect that you'll get a reliable 512Kbps upstream. VoIP is sensitive to loss, latency, and jitter. You may be able, for example, to only get 384Kbps reliably out of the link (before packet loss/jitter/etc wreck its suitability for VoIP). That's a good time to look seriously at a gateway package like pfSense that can prioritize certain classes of traffic while also limiting overall bandwidth. As an example, we noticed on the local business cable offering (2Mbps up) Shaped PL min avg max stddev 2.2M3 6.4 251 557 176 2.1M1 7.8 350 584 134 2.0M3 6.4 271 535 132 1.9M1 7 254 527 131 1.8M0 6 79 339 90 1.75M 0 5.9 14 92 11 1.7M0 5.4 13 77 10 1.65M 0 4.9 11 69 7 1.6M0 5.4 13 55 9 1.5M0 5.3 11 59 7 1.4M0 5 11 57 7 1.3M0 4.9 11 54 6 1.2M0 4.9 11 52 7 1.1M0 4.8 14 53 11 The max starts trending up after 1.6M (helps to graph it) and pretty much everything goes to hell after 1.75M. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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 on a Beagleboard?
> > Out of curiosity, has someone managed to run Asterisk on a Beagleboard > > for home-use? > > > > www.beagleboard.org > > > > As an alternative to a PC, it can be powered from a USB hub, so that > > would make for a compact, fanless Asterisk server. > > > > Thank you. > > 128m of ram & 256 m flash for the 'hard drive' is not much in either > catagory. And ethernet is a USB addon, not on the board. It appears to support a SD/MMC card, meaning that it can support gigs of low power storage space. Or a USB HDD for higher power storage space. 128m of RAM isn't a lot, but some people are apparently running Asterisk on 32MB Linksys WRT54GS's (OpenWRT). If you were careful and cautious, it'd probably work. The Ethernet as an add-on kinda stinks and is probably the largest negative. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] how to sniff RTP and SIP traffic only
Wireshark will support this... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Xavier Cardil Sent: Monday, June 29, 2009 5:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to sniff RTP and SIP traffic only Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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@lists.digium.com
Make sure this is in your xml config: ip.of.ntp.server Unicast On Thu, Jul 2, 2009 at 10:27 AM, mahboob zaman wrote: > > > -- Forwarded message -- > From: mahboob zaman > Date: Tue, Jun 30, 2009 at 8:42 AM > Subject: cisco phone 7911 & > To: asterisk-users@lists.digium.com > > > Hellow, > > I have cisco 7911 and 7906 worked with asterisk server. But i can not set > the time and date for these phones. can any one tell me how can i set the > time and date for these phone. > > Thanks > mahboob > > > -- > Mahboob Zaman > System Engr > Systems & Services Limited > Cell: +8801712280308 > > ___ > -- 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] Compact, fanless appliance?
> On 23 Apr 2009, at 11:34, Vincent wrote: > > Hello, > > On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco > > wrote: > >> Can you give us some clues as to why you have disqualified the > >> fanless > >> and/or embedded devices that are normally recommended on the list > >> (Soekris, etc)? > > > > I haven't: I'd like to know what the options are. I'm looking for an > > up-to-date list of commercially-available compact solutions to run > > Asterisk, including those from Soekris, Atcom, etc. > > http://tinyurl.com/df8qfm Oh, thanks for that. I would also suggest http://tinyurl.com/d5nr8n http://tinyurl.com/ckp4pd ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] Compact, fanless appliance?
> Hello > > For those SOHO customers (ie. at most, a couple of POTS/ISDN > connections and simultaneous SIP calls) who'd rather not use a big, > noisy PC to run Asterisk, I'd like to offer an alternative that has > the following features: > > - not old hardware sold on eBay, ie. it must be up-to-date hardware > sold by a company currently in business > - compact, silent > - has room for a 2.5" hard-disk, but if not, must provide a > CompactFlash plug > - ideally, room for a PCI card, possibly laid down with a riser to > save space > - total cost (shipping + VAT) < 200 euro > > If it's cheaper and not much work, I don't mind buying the parts and > putting the box together myself, but otherwise, I'd rather order a > complete box, ready-to-use. > > What are my options to provide customers with that kind of solution? > > Thank you for any hint. Can you give us some clues as to why you have disqualified the fanless and/or embedded devices that are normally recommended on the list (Soekris, etc)? ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] Special Information Tones
> > On Fri, 20 Mar 2009, Stephen Davies wrote: > > > Hi, > > > > Are you sure that Verizon amswers the call? They should play that > > message as 'early media' without answering, after which they cam clear > > the call with an appropriate cause code. > > Similar issue in the UK and yes, the carriers do answer the call - because > from that second onward thy are taking revenue. > > BT offer a free voicemailbox on landlines too - for the same reason. Many carriers allow you to opt out of these sorts of misfeatures, though you may have to be somewhat insistent. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] USA BRI -- any hope at all?
> I'm quite happy to share what we've done with anyone who is interested > in our solution. By the way, I'm also willing to work with any developers who want to do US BRI work. The Adtran ports can play as either NT or TE, so it is trivial to hook up a card to a spare port and pretend to be a CO for dev work, and we have real BRI's available for testing as well. E-mail me directly if interested. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] USA BRI -- any hope at all?
> >You might want to look into Cisco hardware, their WIC-1B-U cards work =20 > fine in the US, or they did 10 years ago when I last used them for =20 > VoIP. No, you didn't. :-) You might have used a VIC-2BRI-S/T-TE or one of the other voice cards... but the WIC cards are WAN Interface Cards. If only it were that easy. > Used the WIC-1B-U is going for under $50 on eBay. An old 1600 =20 > or 1700 series router with an IOS that supports SIP wouldn't cost much =20 > either. I can probably dig up such a setup here if anyone wants one, except it won't do VoIP. > Help me connect the dots here. I indeed see WIC-1B-U cards and 1721 = > routers. It looks like a pair could be purchased for probably $25. How = Yeah, that won't work... > does that fit into an Asterisk system? I can see how it would be used = > for 128K data, but how does Asterisk pick up and manipulate the data? = > Call setup? Call answering? Does the 1721 deliver VoIP data that = > Asterisk can process? Does Asterisk have a channel interface that can = > accept and use this? Asterisk would deal with it just as it would any other SIP-based service adapter, I would imagine. Imagine it as a large Sipura 3000, or whatever the contemporary VOIP-to-POTS gateway of the modern era is. Have a look at the compatibility chart at http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080111b16.shtml This will give you an idea about combinations that ought to work, I believe. Unfortunately, I don't know if any of them are cheap. We've had BRI for over a decade. It rocks. It's incredibly flexible and is not susceptible to the radio stations that are broadcasting at 50 billion gigawatts half a mile from here (POTS goes to hell). I can do fun stuff like terminating calls into a USR I-Modem, giving me the ability to do 56k remote dialup access into our local network, etc. Unfortunately, getting it to work with Asterisk is a bit of a boggle. We had considered going the Cisco route. All things considered, that has a lot going for it, but the last time I evaluated that solution, it was expensive and the configuration looked somewhat complicated, and I wasn't confident I could manage it without assistance. The other workable solution that I'm aware of seems to involve the Eicon Diva Server boards, which I've heard work. We're currently using an Adtran Atlas 550 to convert the BRI to T1 or PRI or something that Asterisk *can* handle, and this works pretty well. There are some caveats, however, such as the fact that we can not get the Adtran to originate outgoing calls with the second DN on a BRI line (incoming is just fine, though). If anyone goes the Cisco route and is willing to share information on how that worked, and configuration snippets, let me know. I'm quite happy to share what we've done with anyone who is interested in our solution. As for the people who are claiming that BRI is dead, well, not quite. I had called SBC or Ameritech or whatever they were called that day several years ago for what appeared to be a loop problem, and had gotten the "oh I know who you need to talk to" runaround of about twelve departments. However, recently, our Adtran 550 had baked something in its configuration, and we lost one of the BRI's. Not suspecting a problem with the 550, having rebooted it and having eyeballed the config and logs, I dialed with dread the new customer service number for AT&T (which is the number they hand out for POTS repair). I was dumbfounded to navigate the automated system and then to talk immediately to a gentleman who not only told me that the line was a BRI, but also that it appeared to be fine but that they weren't seeing a D channel. Thinking it might be a fried port on the Adtran, I reconfigured to a different port, and it worked. Doh! So then I cleared the configuration on the old port (verifying that it *also* looked correct) and re-entered it, and ... it worked. Double doh! So, if anything, the support situation for BRI's has improved, at least for us here. I expect that the number of new installs is not at an all-time high or anything. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] 'Asterisk is not thread safe' message
I recently built Asterisk from scratch on Ubuntu (Ubuntu 4.2.3-2ubuntu7). Everything seemed to build ok, but when I start Asterisk, I get the message: Warning! Asterisk is not thread safe. Is this anything to be concerned about? How can I make it go away? Is there an alternative threading library I can link against? Thanks! Joe ___ -- 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] ISDN
> On Mon, 13 Oct 2008, Steve Totaro wrote: > > I have done this. Why BRIs exist in the US is beyond me. If you can, > > don't go with BRI. > > Why didn't BRI catch-on in the US? Stupidity. Okay, well, many reasons. It was targetted as a business service, and the pricing models (at least locally) didn't offer a reasonable residential offering until ... I'm thinking almost the mid 1990's (1994-1995). That's when I first recall hearing about the availability of residential BRI that didn't charge by the minute, I think. The telcos had this huge amount of legacy plant installed, much of which wasn't particularly great copper. Good enough for voice. There was also lots of CO and remote equipment for which BRI simply wasn't available. I fondly remember Ameritech running BRI lines out of MILWWI10, using a bunch of repeaters, because they didn't have the capabilities (or capacity, can't recall for sure) at MILWWI45, (you know what's interesting, Google Maps knows CLLI codes!) which resulted in about 11 extra miles on a BRI circuit, and I guess it was much worse in wire miles. It's useful to remember that the phone company expects their equipment to be good for decades, and so there's a huge amount of resistance to upgrading "just for newfangled data services." Now, I know that there was this huge "vision" in the '60's and '70's of the possibility of things like videophones, and ISDN might well have been the ideal platform for delivering something like that, but the flip side is that the Carterfone decision resulted in a booming non-Ma-Bell CPE business and Ma Bell mostly realizing that they were getting shut out of that market. Now, prior to that point, you had a situation where it was Only The Phone Guy who would bring CPE to your house and hook it up, you might dare to move the phone line or jack, but usually not. This meant that a service that was more complex to provision would still be relatively easy for Ma Bell to deploy, since it was just training and equipping their techs that was important. The end user would have had no idea what the underlying technology was. After that point, though, it would have been really hard to sell a residential BRI, unless you had equipment capable of automatic configuration, because it's hard enough for Joe Sixpack to plug in a POTS line correctly as it is. Configuring more-complex stuff, especially in the days before nifty little GUI interfaces (which requires electronic capabilities not really present until recent years) would have been rough. Despite all this, there was a renaissance with BRI in the 1990's. We had reached a point where the electronics were reasonable. The Internet did not yet exist for most people, and modem technology was 9600 or 14.4. As corporate networking and the Internet exploded, there was a willingness to pay premium prices for ISDN gear that would allow relatively inexpensive BRI circuits to attach at speeds far beyond POTS. Then we saw that fall apart, as speeds increased to 28.8 and then 56K, and for most users, that was close enough. DSL was right on the heels of that, offering greater-than-ISDN speeds. ISDN BRI was back to dead status by about 2000. You can *see* this in terms of CPE devices that supported ISDN BRI. In the meantime, the ILEC's began to truly understand the difference between switched circuit and packet data services. Many people had been using ISDN BRI as a faster and more flexible alternative to 56K DDS lines, which tended to tie up switch capacity. More people were ordering "second lines," and leaving them connected to local ISP's for hours at a time, which created trunk, network, and switch capacity challenges for the ILEC's. This was devastating to the ILEC's, which typically plan capital expenditures to be good for many years, but in this case, I have to assume that the ILEC planners knew that Internet would be provided over circuits other than their switched POTS/DS0's in the near future, which would dump capacity requirements back down. They even got smacked harder than maybe they expected, as some people gave up land lines entirely, in favor of cell... In the meantime, many of the major "data" uses that had been envisioned for ISDN BRI have been done, better, cheaper, on the Internet. Videophone? Easy on a PC with speakers, mic, and an Internet connection, but hopelessly challenging to someone with a POTS or BRI line. Private network interconnection? VPN over Internet. Etc. This has meant that BRI has "evolved" towards a way to deliver telephone network with no loss, or, rather, most of the envisioned benefits are no longer likely to be exploited via BRI. So you don't see many BRI lines, and it is pretty common for those that you do see to be hooked up to a PBX, automated call ha
Re: [asterisk-users] ISDN
> >With ISDN, the conversion is done in your phone > > Exactly. Or in the case of Asterisk, it is a 4 wire digital right into = > the switch--no degradation. Even converting back and forth between = > analog and digital multiple times compromises quality. Try doing a = > dial-up modem across such a path. The best you will get is 20 - 30 K. A single D/A hop destroys the ability to do 56k. Successful 56k requires that there be a single A/D hop at the far end (the user's POTS interface) and then digital delivery of signal the remainder of the way to the terminating equipment. (modem -> phone co A/D -> digital to ISP modem bank). If you stick an extra D/A (maybe plus A/D) transformation in there, you will probably get fairly clean speeds in the upper ranges of 28.8-33.6, but that'll be it (modem -> phone co A/D -> digital network -> phone co D/A -> your buddy's modem). If you're unlucky enough to get some crummy phone co arrangement where they punt you back and forth from digital to analog and back to digital within their network, that's even worse. >From an Asterisk point of view, it's interesting that you can get digital delivery of the signal, and route the signal around internally digitally, if you have ISDN. This means, for example, that our USR Courier I-Modem, which can terminate a 56K call *digitally*, results in my being able to make a 56K connection from most modern cities here in the US, without wasting an ILEC ISDN BRI line dedicated to that purpose, by having the PBX connect an extension to the I-Modem. I just dial into a general purpose number, and dial the appropriate extension, and voila, I'm on our network at "high speed." This is clearly obvious to you, but I thought I'd expand for the others who might be reading along and didn't understand the implications of all- digital. > >IF you can get a PRI-line for the same price. > > Not to mention that the interfaces for PRI are about five times as = > expensive. I'm not sure why. It doesn't seem like it ought to take a = > lot of electronics to break down the bit stream. It may not take a lot of electronics. However, the sad truth of it all is that any electronic device produced at low volume tends to be expensive to produce. This is largely the result of costs such as retooling, and in most cases the significantly higher cost of small-run integrated circuits. For example, a PC board manufacturing house (I'll use the following shop, no affiliation, as an example, because they have transparent pricing) http://www.expresspcb.com/ExpressPCBHtm/Specs4LayerStandard.htm http://www.expresspcb.com/ExpressPCBHtm/Specs4LayerProduction.htm for a 30 sq in board. To produce 10 boards would cost $404, or $40/board. To produce 50 boards would cost $1109, or $22/board. To produce 1000 boards would cost $15516, or $15/board. Even 1000 isn't really a large run, though. You're paying premium board rates for small runs, because the shop has to stop and retool for your run. I haven't bothered to get a large-run quote, but I bet you can get that down to well under $10/board if you're ordering a hundred thousand at a time... You then have to add on assembly costs, which are typically higher than the PCB costs. It could very easily end up costing $50/board *just* for PCB and assembly, no parts included, for runs in the hundreds of cards range. The problem with telephony stuff, especially in this market, will be that the demand for a T1(/PRI/etc) interface is going to be very low. You would need to be a relatively big shop to be able to buy by the thousand, as even at one bulk buy per year, that translates to several cards departing inventory daily. I expect that some of the ISDN BRI interfaces are dirt cheap because they're popular over in Europe. I've been told that in many places, they're sold in lieu of a modem. Once you are moving product in high volumes, the pricing tends to come down. It stinks, yes. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] ISDN
> I'm in the process of setting up Asterisk in a SOHO environment using = > ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID = > is used for the business and the other is used for personal. The = > circuit already exists, but is presently being interfaced to POTS phones = > via a TA. > > This configuration is not very common in the US, but we are fortunate = > that our LEC offers it price competitively with equivalent POTS services = > and it makes more sense, both in terms of voice quality (4 wire digital = > to the PABX) and flexibility. > > Ideally it would allow any combination of two calls, identified by SPID. > > If anyone has done anything similar, or has any experience with BRI = > ISDN, I would appreciate input and direction. > > If anyone knows where documentation exists on configuring ISDN, that = > information would also be greatly appreciated. Asterisk has a bit of a = > learning curve, and ISDN BRI isn't the most widely used or covered = > aspect of it. BTW, I have a strong telecom background, so the theory = > part of it will not be a problem, only the necessary documentation to = > apply it to Asterisk. The one solution I've heard, on and off again, that works with Asterisk here in the US is the Eicon Diva cards. There are other solutions. Where I am, we're unreasonably close to a local radio conglomerate that has a number of high power antennas. We found early on that RF interference was a killer, which caused me to run a lot of our telecom and data wiring in conduit. Unfortunately, we discovered that POTS lines were a hell of a mess when connected to anything more complex than a phone or two. Lots of RF interference. Church radio music on Sundays, even. So, we brought our lines in on BRI, which we've used for data and voice elsewhere. Being eternally frustrated with the lack of ISDN support after maybe 2000 here in the US (we have a bunch of interesting ISDN gear from the 90's!), I set out to see what I could do to interface BRI to Asterisk. I *didn't* go the Eicon route, because at the time it was considered relatively unreliable. Instead, we picked up an Adtran Atlas 550, which can handle ISDN BRI, PRI, POTS, etc. We have been using the Atlas as a translator to convert BRI to T1, which works moderately well, but we've seen some issues, mostly in the capabilities of the Adtran (such as an inability to select the desired SPID/DN for outgoing calls). The Adtran has some other amazing capabilities, such as providing FXO/FXS ports, and even ISDN BRI ports for other devices we'd liked hooked into our PBX. Despite that, I'd love to see an ISDN BRI solution for the US. I might be willing to test the Eicon Diva Server card... hm. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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 add Callee's name into Dial command ?
I think this is what you want: http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier <[EMAIL PROTECTED]> wrote:
> Hi,
>
> When dialing a number, I use :
> exten => _123X, 1, Dial (SIP/${EXTEN})
>
> Then, I get TRYING and RINGING SIP messages which both include this kind of
> line :
> To:
>
> Is it possible, configuring Asterisk 1.4, to get something like this
> instead ?
> To: "John Doe"
>
> This way, I'm hoping to display callee's name beside (or instead of)
> callee's number which would offer a double check for caller which might be
> confusing extensions, for instance.
>
>
> I tried this :
> exten => _123X, 1, SIPAddHeader(To: Doe \ \;user=phone\>)
>
> but I still got :
> To:
>
> Regards
>
> ___
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[asterisk-users] Read Command
I've search the world over but I haven't figured out a way to have valid/invalid options for entry when using the Read command... I need to set a variable, but only want to allow certain values to be valid options for that variable... Any ideas? Thanks in advance.. -JC ___ -- 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] Any chance this is related to fastagi: received mini-frame before full voice frame
Hi, I have recently been having difficulty with cmd record where calls are not being recorded. I would like to know whether it is possible that my fastagi script is the root cause of the problem. I am using a fastagi script written in python to answer the calls, and the dialogue interaction seems to work fine, the call is successfully answered, and I can always hear the audio prompts, however recently I have been seeing a high incidence of failure to record the user generated audio, which is invariably accompanied by iax2 debug messages: "received mini-frame before full voice frame". Another list user kindly explained to me in a previous message that this error message means that, if the first full voice frame never comes through, there is no data on which to base the changes being conveyed in the mini frames, and thus record doesn't have any way of knowing what to record. My understanding of agi is that it simply passes text commands back-and-forth between asterisk and the agi/fastagi/eagi script, over stdin. This would seem to imply that the record failures I've described can't be related to the agi script as the actual recording is not done there anyway. If possible I'd like to rule out agi as the culprit categorically in an effort to reduce the problem space a bit. 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
Re: [asterisk-users] cmdRecord issue related to "iax2 received mini frame before first full voice frame"?
>> A more detailed explanation of what the above warning means/implies, >> and how or why it might be preventing recordings would be greatly >> appreciated. >A mini frame is simply a frame containing minimal information about the >call itself (the meta data), and a full frame contains all of the meta-data >information. Sending mini-frames is part of the IAX protocol, as a way of >saving significant bandwidth over the course of a call. However, a mini frame >cannot be interpreted correctly independently of a full frame. In every media >stream, a full frame is send approximately once every 60 seconds, to sync the >timestamps. >> I'm running Asterisk 1.4.11 on debian Etch. >There have been many changes, bug fixes, and even security issues fixed since >1.4.11. I'd really recommend that you try something more recent (and even the >latest, because we fixed 2 security issues in the latest release, 1.4.21.2). Thank you very much for this clear explanation of what is going on. The implication then is that this issue could indeed be having a significant impact on whether calls are eventually recorded? I plan to look into upgrading our distribution, but this appears to be impacting 1/10 calls coming through our system at the moment so I wonder if there is some/any way to help make sure that that first full voice frame gets sent before the miniframes get pushed through? I also wonder if there is a point after which the call is lost, regardless of whether the first full voice frame comes through? could other issues like mismatches between the system clocks on the respective machines also be impacting performance? thanks a bunch! -- Tilghman ___ -- 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] cmdRecord issue related to "iax2 received mini frame before first full voice frame"?
Hi, I tried sending this message a few months back but never lucked into a response. I thought I'd try one more time, juicing up the subject heading a bit, as I am still seeing this behavior intermittently. I'm running several asterisk servers in combination with dundi. The servers are in different data centers, but other than that they are running identical copies of the same os image, asterisk configuration, etc. One server acts as the trunk and is used to terminate pstn calls, and pass them on to another server via dundi, which then answers the call. I recently noticed that one of the call-answering servers was responding and playing back voice prompts fine, but was failing to record any user generated audio. After opening up the CLI on this server and running a test call through it, I noticed reams of the following warning message any time audio was being played or recorded: [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame I found the following related post: http://lists.digium.com/pipermail/asterisk-users/2006-January/136982.html However this doesn't explain why I should be unable to record anything. The issue seems to be related to network activity, and I'm not seeing it on any other servers. A more detailed explanation of what the above warning means/implies, and how or why it might be preventing recordings would be greatly appreciated. I'm running Asterisk 1.4.11 on debian Etch. Cheers ___ -- 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] Remote Support
Does anyone have any suggestions on what to use to monitor a vendor doing remote support? On the windows side things are typically done via screen sharing ( gotoassist.com, bomgar or similar) so at least you can see what the other end is doing. In working with linux (especially hardware vendors for asterisk) they want ssh root access, but I'm nervous about giving someone free range to a box without any type of monitoring. What does everyone else do? (besides not give them access). Looking at something like screen sharing or recording, perhaps keystroke logging. ___ -- 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] Magnetic door locks
> One of the last "secure" facilities I worked in had a motion sensor > that unlocked the door for people leaving from one door. The COO was > pretty shocked when I took off my belt, easily pushed it through the > gap in the glass doors near the top and triggered the motion sensor, > immediately opening the door to the admin side of the building. > > The other side did not have motion sensor just an RFID and magnetic > lock. Under challenge, I grabbed a chair, stood on it, lifted the > drop ceiling and cut one of the wires to the magnet, click, it opened > (would be a fire hazard if door did not unlock when power was > removed). > > I think they spent big bucks on their false "sense of security" I also > pointed out that the data room could be accessed via drop ceiling on > one side and drywall (that's tough stuff!!) on two of the three walls. You have to remember that many "secure" facilities were not designed by security professionals, but are instead simply areas where a company decided that they wanted to be secure long after a building was built. They call a contractor, ask for a wall, and instant "secure" area. Then the security company comes in, company says "we want restricted access," and voila, you get a nice doorstrike and an RFID reader. Security is never absolute. You had a situation that was likely to keep typical people out of the "secured" area. Fix those problems and then there are others. A good crowbar is effective against a *lot* of door hardware. What, you have a nice steel door? Did you know that you can often bend the door a bit and pop the latch? Oh, you've got a latch protector? Great, even more leverage for the crowbar... etc... And walls. Sure, a drywall wall is pretty flimsy stuff, but a sledgehammer can make pretty decent work out of a cinderblock wall... You pretty much have to figure out what level of security you actually want to obtain. Putting security system wires in conduit is better than just drilling around with a bellhanger, but would it necessarily have prevented you from finding a junction box and getting at the magnet wires? It helps to have a facility built from the ground up to be secure. And even many of those don't do many of the things that they could. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] MagicJack quality
> I am puzzled by the quality of magicjack. I keep trying to figure out how > they can the quality be that adequate. Since Skype also has an excellent > quality, that leaves me to believe that software based calls (softphones) > could have and advantage over hardphones, provided there is a parameter that > those 2 companies are addressing. You are puzzled by the quality? http://www.laptopmag.com/review/voip/magicjack.aspx I don't know, but from the sounds of the comments, you'd get about just as much quality out of an actual cigarette lighter, and probably a good bit more usefulness. Nice EULA, by the way: http://gadgets.boingboing.net/2008/04/14/magicjacks-eula-says.html VoIP over the Internet isn't /that/ hard. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- 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] US T1 Hangup Detection
> On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
> > > Really? You have an RJ-21X block that contains both analog AND T1
> > > wires? That's really uncommon. I hope they at least put the red
> > > special service caps on the T1 wires.
> >
> > Yup. I thought that pretty funny myself. 10 year old analog wires
> > running a digital T1. :) And they do have some caps on them, I think
> > it was red but not 100% sure.
>
> No, that's not the unusual part. The unusual part is just that both
> analog and digital services are on the same block. Maybe it's a
> regional think...
That's really not unusual. It's not /preferred/, but that's an entirely
different can of worms.
In general, if copper is available into a building, the telco is going to
look very seriously at the possibility of using that. If the building is
already wired and the copper tests clean, the telco will want to use that.
In most existing situations, that will already be terminated in a can with
lightning suppression and will have been crossed over to RJ21X's that are
going to whatever suites are in the building.
Since the telco will have /no/ /problem/ running the T1 over their outside
plant and up to the can on what is approximately Category 3 wire, and the
T1 signal is going to have been running alongside those same "analog wires"
for probably a few miles, what happens next should be obvious.
Suite 214 wants a T1. There's already a 25-pair going up there from the
RJ21X. It's second story, so do you go and spend an {hour, afternoon,
etc} figuring out how to run fresh wire, or do you notice that only 6 pair
are in use on the RJ21X, and decide to feed up on the existing cable?
Now, if you're nasty and you don't separate it (typically I see the bottom
used for data) and you don't put redcaps on, yeah, then that is just
looking for eventual trouble. And who knows, the wire may be cruddy, so
maybe you still end up doing the separate run. But it probably works.
I've seen this often enough. Would I prefer to see new cable run? Sure.
But we've all done our copper sins. I've seen a lot of things that are
uglier than that. Here's one of them:
http://www.sol.net/hallofshame/
(I've always meant to expand that page, but it seems that I never get the
good photos of bad stuff)
Lack of space, lack of need, lack of having another RJ21X in the truck are
just a few other obvious reasons that this might be done.
... JG
--
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.
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[asterisk-users] Hangup?
I've got a unique situation and think it may be the lack of the Hangup command in the dialplan that is creating the issue.Can anyone elaborate on why it is, or is not, important to use hangup in the dialplan. Presently I don't have the first instance of it in my dialplan, however, I see some things in the debugging that might be cleaner if I did implement hangup I have approximately 140 extensions provisioned off this asterisk server and about 8 IVRs...So as you might expect, it is quite busy... Thanks, -Joe ___ -- 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] SIP vs. SKINNY
Can anyone comment on the performance benefits when comparing sip to skinny ? ___ -- 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] need ata suggestion
Grandstream 4004 will do this... -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Tuesday, June 17, 2008 1:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] need ata suggestion On Mon, 16 Jun 2008, Eric Fort wrote: > I'm presently working on provisioning VoIP to a traditional key system. I > have a single SIP DID inbound that gives me a maximum of 2 concurrent > channels. I need an ATA that will ring the second station port when the > first is in use. What devices will do this with a single sip registration > with the provider? Er, Asterisk will do this with a TDM400 card or clone. Expensive ATA though :) But maybe an AVM Fritz! box will work for you too... Gordon ___ -- 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] T.38 Passthru w/ MediaGateway | Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN
Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN ___ -- 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] 911 via MAX TNT ??
Any suggestions ?
Available options for the two settings similiar to the one identified are as
follows:
admin> set send-dnis-type-of-number?
send-dnis-type-of-number:
Type of Number to be sent in called party IE in the setup message to
pstn. For ISDN signaling. To be used on egress gateway for VoIP calls.
Enumerated field, values:
unknown:
international:
national:
network-spec:
subscriber:
abbreviated:
transparent: Setting this, we can pass TON transparently as received from
upper layers or in case of VoIP, as received from Near End gateway.
admin> set send-dnis-numbering-plan?
send-dnis-numbering-plan:
Numbering Plan to be sent in called party IE in the setup message to
pstn. For ISDN signaling. To be used on egress gateway for VoIP calls.
Enumerated field, values:
unknown:
isdn-telephony:
data:
telex:
national:
private:
transparent: Setting this, we can pass NP transparently as received from
upper layers or in case of VoIP, as received from Near End gateway.
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Leon Sun [EMAIL
PROTECTED]
Sent: Monday, June 09, 2008 1:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
It should work.
Leon Sun
Times Telecom
Tel: 604-279-8787 ext 1586
Fax: 604-278-2793
Mobile: 604-780-2668
Click this button now and leave your phone number. Talk to me for free.
powered by www.clicksaya.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP.. Would this apply
in that scenario ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk
group / route.
Joe Carroll wrote:
> We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???
>
> PRI DEBUG FOLLOWS:
>
>
> <--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08
> Bearer_Cap 80 90 A2 (Speech,Rate=64K)
> Channel_Id A1 83 83 (Pref,Intf=0,Chan=3)
> Calling_Num (National,Restricted,Failed) 229317
> Called_Num (National) 911
>
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
> Sent: Thursday, June 05, 2008 6:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
>
> Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that
a carrier would deliver to an end user...
>
>
>
>
> From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
> Sent: Thursday, June 05, 2008 9:27 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
>
> On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
>> On June 4, 2008 06:20:57 pm Joe Carroll wrote:
>>> Interestingly enough, on the syslog messages from the TNT we are seeing
>>> "Called = 911, Q850 Cause = 28, SIP Response = 484"
>> That really looks like the switch that the TNT is talking to is rejec
[asterisk-users] Echo on PRI even with H/W echo cancel
Hello, I have a PRI coming into a Digium TE122B with hardware echo cancel, but we are still experiencing echo on the first 10 seconds of a call. Is there anything that can be done about this? I have tried contacting digium support, but have not heard back from them (placed a support incident about a week ago). I see on digiums website that some of their card have a VPMOCT128 Octasic echo cancel, but the TE122B comes with digiums VPMADT032 echo canceler. Is the octastic echo cancel better? Should I look into a card with Octastic echo canceler? I see Sangoma has a single port with T1 with Octastic echo canceler or would have to move up to the dual span card to get the Octastic echo cancel on the digium card. ___ -- 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] ?
___ -- 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] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP.. Would this apply in
that scenario ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk
group / route.
Joe Carroll wrote:
> We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???
>
> PRI DEBUG FOLLOWS:
>
>
> <--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08
> Bearer_Cap 80 90 A2 (Speech,Rate=64K)
> Channel_Id A1 83 83 (Pref,Intf=0,Chan=3)
> Calling_Num (National,Restricted,Failed) 229317
> Called_Num (National) 911
>
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
> Sent: Thursday, June 05, 2008 6:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
>
> Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that
a carrier would deliver to an end user...
>
>
>
>
> From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
> Sent: Thursday, June 05, 2008 9:27 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] 911 via MAX TNT ??
>
> On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
>> On June 4, 2008 06:20:57 pm Joe Carroll wrote:
>>> Interestingly enough, on the syslog messages from the TNT we are seeing
>>> "Called = 911, Q850 Cause = 28, SIP Response = 484"
>> That really looks like the switch that the TNT is talking to is rejecting
the
>> number, not the TNT...
>
> Remember: "9-1-1" is a *dialling pattern*, not a *directory number*;
> it's entirely possible that trunks wouldn't accept it directly.
>
> This *is* a *LEC* trunk, right?
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth Baylink
[EMAIL PROTECTED]
> Designer The Things I Think RFC
2100
> Ashworth & Associates http://baylink.pitas.com '87
e24
> St Petersburg FL USA http://photo.imageinc.us +1 727 647
1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
>-- (Joseph Stalin)
>
> ___
> -- 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
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678
Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider sending a sample max tnt config reflecting how this is done? Thank you.. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, June 06, 2008 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? I believe the ISDN call plan can be configured as part of the trunk group / route. Joe Carroll wrote: > We talked with the LEC and discovered that 911 has to be sent as Unknown > instead of National... Anyone know how we might tell the TNT to do this? > Apparently, according to the carrier, all Special Access Numbers, 411, 611, > 911, etc require this special code ??? > > PRI DEBUG FOLLOWS: > > > <--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 > Bearer_Cap 80 90 A2 (Speech,Rate=64K) > Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) > Calling_Num (National,Restricted,Failed) 229317 > Called_Num (National) 911 > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll > Sent: Thursday, June 05, 2008 6:52 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] 911 via MAX TNT ?? > > Yes, we are using the max tnt to aggregate several PRIs both inbound and > outbound from multiple carriers. This PRI is a normal two way circuit that a > carrier would deliver to an end user... > > > > > From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL > PROTECTED] > Sent: Thursday, June 05, 2008 9:27 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] 911 via MAX TNT ?? > > On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: >> On June 4, 2008 06:20:57 pm Joe Carroll wrote: >>> Interestingly enough, on the syslog messages from the TNT we are seeing >>> "Called = 911, Q850 Cause = 28, SIP Response = 484" >> That really looks like the switch that the TNT is talking to is rejecting the >> number, not the TNT... > > Remember: "9-1-1" is a *dialling pattern*, not a *directory number*; > it's entirely possible that trunks wouldn't accept it directly. > > This *is* a *LEC* trunk, right? > > Cheers, > -- jra > -- > Jay R. Ashworth Baylink [EMAIL > PROTECTED] > Designer The Things I Think RFC 2100 > Ashworth & Associates http://baylink.pitas.com '87 e24 > St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 > > Those who cast the vote decide nothing. > Those who count the vote decide everything. >-- (Joseph Stalin) > > ___ > -- 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 -- 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 -- 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] 911 via MAX TNT ??
We talked with the LEC and discovered that 911 has to be sent as Unknown instead of National... Anyone know how we might tell the TNT to do this? Apparently, according to the carrier, all Special Access Numbers, 411, 611, 911, etc require this special code ??? PRI DEBUG FOLLOWS: <--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08 Bearer_Cap 80 90 A2 (Speech,Rate=64K) Channel_Id A1 83 83 (Pref,Intf=0,Chan=3) Calling_Num (National,Restricted,Failed) 229317 Called_Num (National) 911 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Thursday, June 05, 2008 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 via MAX TNT ?? Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: > On June 4, 2008 06:20:57 pm Joe Carroll wrote: > > Interestingly enough, on the syslog messages from the TNT we are seeing > > "Called = 911, Q850 Cause = 28, SIP Response = 484" > > That really looks like the switch that the TNT is talking to is rejecting the > number, not the TNT... Remember: "9-1-1" is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Similar extension numbers for multiple users
Would it be possible to have a context with includes for each tenant and include that context in the specific tenant contexts that you would have calling each other.if that makes any sense whatsoever.. From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Carlos Chavez [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Similar extension numbers for multiple users As long as each tenant has its own context you can use the same numbering plan. The only thing you need to keep unique are the names for the SIP devices. If you want your tenants to be able to call each other then you would need to set up a special prefix for each tenant. On Thu, 2008-06-05 at 18:01 -0400, Zeeshan Zakaria wrote: > Hi everybody, > > Is it possible to create similar extension numbers for multiple users. > I am looking at a case of virtual PBX with 5 tenants on one server. > Any applicable ideas or suggestions would be highly appreciated. > > -- > Zeeshan A Zakaria > ___ > -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnologìa +52-55-91169161 ext 2001 ___ -- 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] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and outbound from multiple carriers. This PRI is a normal two way circuit that a carrier would deliver to an end user... From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth [EMAIL PROTECTED] Sent: Thursday, June 05, 2008 9:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 911 via MAX TNT ?? On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote: > On June 4, 2008 06:20:57 pm Joe Carroll wrote: > > Interestingly enough, on the syslog messages from the TNT we are seeing > > "Called = 911, Q850 Cause = 28, SIP Response = 484" > > That really looks like the switch that the TNT is talking to is rejecting the > number, not the TNT... Remember: "9-1-1" is a *dialling pattern*, not a *directory number*; it's entirely possible that trunks wouldn't accept it directly. This *is* a *LEC* trunk, right? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] 911 via MAX TNT ??
See below, we replaced the area code and prefix of with NPANXX for concerns
Interestingly enough, on the syslog messages from the TNT we are seeing "Called
= 911, Q850 Cause = 28, SIP Response = 484"
Extension Changed NPANXX7604 new state InUse for Notify User NPANXX7555
-- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518",
"CALLERID(number)=NPANXX3551") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/NPANXX7604-08c46518",
"SIP/To-TNT/3100911") in new stack
-- Called To-TNT/3100911
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY
-- Got SIP response 484 "Address Incomplete" back from 172.16.10.230
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/NPANXX7604-08c46518' status is 'CHANUNAVAIL'
-- Executing [EMAIL PROTECTED]:1] Set("SIP/NPANXX7604-08c46518",
"CDR(userfield)=") in new stack
Extension Changed NXX5557604 new state Idle for Notify User NXX5557555
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez
Sent: Wednesday, June 04, 2008 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
The first place you may want to look is in the SYSLOG of the TNT,
allowing you to see things such as the ISDN error code along with the
SIP code. You can try to catch that on the terminal of the TNT, but it
may make more sense to pipe your syslogs out to an external box, if you
aren't doing it already.
JR's suggestion that it may be a limit of the trunk you're using.
___
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Re: [asterisk-users] End call behaviour
Is reinvite set tp yes for the device? -Original Message- From: "Joseph L. Casale" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: 5/25/08 7:52 PM Subject: Re: [asterisk-users] End call behaviour >What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a >SIP phone ? Hi, These are SIP phones (Snom M3's and Astra 480i's), I didn't notice this when I was testing with my softphone but I cant recall :) Thanks! jlc ___ -- 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] trying directrtpsetup
RTP DEBUG IP from the asterisk CLI From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of ronald ramos [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 4:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] trying directrtpsetup Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron ___ -- 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] End call behaviour
What type endpoint do you have ? Channel bank perhaps ? Is it an ATA ? a SIP phone ? From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Joseph L. Casale [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 11:58 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] End call behaviour When I exit voicemail or an inbound caller hangs up I hear a busy signal for a few seconds before Asterisk terminates the call. I thought this behavior was handled in the dial plan with a Hangup() command? How can I correct this? Thanks, jlc ___ -- 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] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider
In your account settings (sip.conf) for the PAP2T device, do you have reinvite enabled for one or both.. the SIP provider and/or the PAP2T device ? From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Shaun Wingrin [EMAIL PROTECTED] Sent: Sunday, May 25, 2008 3:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider Shaun schrieb: > Hi All, > > This is puzzling me greatly. > > The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to > Asterisk are SIP clients. Codec throughout G729 (only have 1 license on > Asterisk server loaded though). When calling the SIP clients from PAP2T I > can't hear them but they can hear me. > > If I call from PAP2T through Asterisk using?IAX2 to a VOIP provider there is > speach in both directions! > > Any suggestions? > > Thanks Shaun > check your firewall/nat settings. If your setup will work for around 5 minutes after you have rebooted the pap2t then you have to active the nat keep alive and nat mapping service in the pap2t. best regards steve smith DEAR STEVE, THANKS, I DID CHECK THEM AND THEY NEEDED TO BE ACTIVATED. THIS DOES NOT SOLVE THE PROBLE. STILL ONE WAY SPEECH WHEN CALLING PAP2T -->ASTERISK-->SIP DEVICE ATTACHED. HOWEVER PAP2T-->ASTERISK-->SIP PROVIDER WORKS FINE AS WELL AS SIP DEVICE ATTACHED-->ASTERISK-->SIP RPOVIDER. ANY SUGGESTIOPS WELCOME ___ -- 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] forwarding pots lines
There are a couple of companies out there that make 24 port fxo and fxs boxes. If you have some unused fibers you cout do this very reliably with two channel banks... One with fxs ports and the other with fxo ports and t1 media converters. The grand stream solution mentioned in an earlier post does 8 ports, you could get one 4 port model and one 8 port model of fxs and the same of fxo and accomplish your goal rather inexpensively as well. Joe -Original Message- From: "Dennis P. Clark" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: 5/23/08 8:43 AM Subject: Re: [asterisk-users] forwarding pots lines Sorry to jump in on this but I am also interested in this topic. In my scenario I have about 10 POTs lines brought into the front of a facility and the only infrastructure connecting the back of the facility is a 3000ft fiber backhaul. I've been asked to bring the POTs lines to the back of the facility. Are there any ATAs that trunk multiple POTs Lines? Like a multiplexer of some sort. If anyone has any information can you please provide the manufacturer and model of the device? Thank You, Dennis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, May 23, 2008 4:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines On Fri, May 23, 2008 at 10:04 AM, Eric Fort <[EMAIL PROTECTED]> wrote: > will an ata directly connect to another remote ata thus emulating a long > phone cord? also most of the ATA's I've seen drive a phone rather than > accepting a line from the telco. Depending on the reliability needed (is this a way to talk to a girlfriend in another country or a mission-critical business use?) I'd say it's better to pay a small monthly fee to someone like OnSIP.com and use their centrex. If it's because you have the phone lines already installed and need to just use them at certain times, I do think there are FXO devices but I'm not sure they will help. You wouldn't need two asterisk servers at any rate but only one. The phones connect (through a router if need be) to the asterisk at the phone lines + FXO end. /r ___ -- 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] forwarding pots lines
Two grandsreams, a 4008 and a 4108 would inexpensively do this for you. Instructions are on their site. -Original Message- From: "Alex Balashov" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: 5/23/08 1:21 AM Subject: Re: [asterisk-users] forwarding pots lines Eric Fort wrote: > I'm looking for a simple hardware solution where I can connect POTS > lines at one place and make them appear transparently at another > location with only SIP and the internet between the locations. If I'm > thinking this out right one location would need a box with a bunch of > fxo interfaces and the other would need a box with a equal number of fxs > interfaces. I'd like this to essentially emulate a really long piece of > phone wire in as many ways as possible. What hardware should I use and > what is the best way to provision this. I'd prefer to forgo the expense > of 2 full asterisk servers as this seems unnecessary for the application. You can use devices called ATAs (Analogue Telephone Adaptors). They are much cheaper. -- 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 -- 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] T.38 w/ MAX TNT & ASTERISK
12.0.2 I am speculating, and i'm not sure what i'm basing the speculation on, but I believe the problem is with timing between the TNT and the ATA. From: Shane Burrell [EMAIL PROTECTED] Sent: Wednesday, May 21, 2008 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 w/ MAX TNT & ASTERISK What TAOS do you have? On May 21, 2008, at 10:39 AM, JR Richardson wrote: >> We solved our former echo issues... however, as luck would have >> it..Faxing is yet a completely different animal. We >> understand that digium doesn't really support faxing with >> asterisk... HOWEVER...it seems that ATA manufacturers and the >> MAX TNT indicate that fax is supported >> >> Scenario: >> >> TDM/PRIs -->MAX TNT <-sip-> asterisk <-sip-> ata ->fax >> >> We should note that the network between the ata and the facility >> with the max and asterisk is a managed fiber network in a type of >> campus environment... 1-2ms latency end to end tops... >> >> If the ata is reinviting to the MAX TNT shouldn't fax work with T. >> 38... Does anyone have any experience with this configuration ? >> Thanks, > > I have been wanting to do this for months, but just can't find the > time to work on it. If you do get it going, I would really appriciate > knowing how. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > > ___ > -- 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] iax2 received mini frame before first full voice frame
Hi, I'm running several asterisk servers in combination with dundi. The servers are in different data centers, but other than that they are running identical copies of the same os image, asterisk configuration, etc. One server acts as the trunk and is used to terminate pstn calls, and pass them on to another server via dundi, which then answers the call. I recently noticed that one of the call-answering servers was responding and playing back voice prompts fine, but was failing to record any user generated audio. After opening up the CLI on this server and running a test call through it, I noticed reams of the following warning message any time audio was being played or recorded: [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: Received mini frame before first full voice frame I found the following related post: http://lists.digium.com/pipermail/asterisk-users/2006-January/136982.html However this doesn't explain why I should be unable to record anything. The issue seems to be related to network activity, and I'm not seeing it on any other servers. A more detailed explanation of what the above warning means/implies, and how or why it might be preventing recordings would be greatly appreciated. I'm running Asterisk 1.4.11 on debian Etch. ___ -- 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] T.38 w/ MAX TNT & ASTERISK
We solved our former echo issues... however, as luck would have it.. Faxing is yet a completely different animal. We understand that digium doesn't really support faxing with asterisk... HOWEVER...it seems that ATA manufacturers and the MAX TNT indicate that fax is supported Scenario: TDM/PRIs -->MAX TNT <-sip-> asterisk <-sip-> ata ->fax We should note that the network between the ata and the facility with the max and asterisk is a managed fiber network in a type of campus environment... 1-2ms latency end to end tops... If the ata is reinviting to the MAX TNT shouldn't fax work with T.38... Does anyone have any experience with this configuration ? Thanks, -J ___ -- 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] Error Counters on PRI Circuit
Is there a way to see error counts on the T1 of a PRI? Hooked up to asterisk via a digium TE122. Looking for something to make sure I'm not getting any CRC, framing or other errors on the T1. Using asterisk 1.4.19 and zaptel 1.4.10 ___ -- 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] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)
Hello... We're attempting to track down an intermittent echo issue. Our setup is sipsippri to carriers. We have less than 2 ms latency on the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is the subscriber using sip. The PSTN users does not hear the echo. We should be note that there is zero echo when calling sip to sip with or without reinvites enabled. We have several different phones; linksys, polycom, & grandstream (both atas and phones). It's difficult to reproduce the problem regularly so isolation is an issue. Thanks in advance.. -Joe ___ -- 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] rebooting newer cisco phones
Does anyone have a solution for remotely getting the newer cisco phones (7941, 7961, 7970, etc ) to reread their configs (or even rebooting). I am running SIP firmware connected to asterisk. Check-sync doesn't seem to work anymore, I can't login to the phones as root because I am given a "challenge: password:" prompt. occasionally I need to make changes to phones at remote locations, only solution I have now is rebooting the POE switch. Kinda an overkill. ___ -- 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] Analog DID
It seems the standard for Analog DID (at least around here) is wink start, does the Rhino cards work with this or do I need to have the telco immediately send the DTMF tones? On Wed, Feb 13, 2008 at 12:33 PM, James Finstrom < [EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Rhino's Analog cards support analog DID. no need for all the extra > stuff You will want to get an R8FXX with fxs modules that will give > you channels in sets of 2. > > ADID has not really taken off in the OS telephony market I think due > to a lack of understanding people stay with the proprietary phone > systems that pimp this feature. Okay so I will take the lead and pimp > it for asterisk. With Rhino Analog cards you CAN do ADID with no extra > equipment. However if you want to spend the money we can go the other > route :) > > darren wrote: > > > > An analog DID trunk is a line (typically part of a group) that has > > a group of numbers assigned to it at the telco side. They work in > > a variety of ways depending on the telco. One example is the > > trunks as Telus provides them. The end user provides dialtone back > > to the telco. When a call comes in on a DID the telco picks up the > > first available line (remember, the customer is providing dial > > tone.) and dials the last 4 digits of the dialed number. They are > > often replaced by PRIs but in some locations a PRI is not > > affordable and these provide the same DID functionality for a small > > fraction of the price. > > > > > > > > Darren Wiebe > > > > [EMAIL PROTECTED] > > > > > > > > > > > > Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to > > asterisk-users@lists.digium.com Subject: Re: [asterisk-users] > > Analog DID > > > > On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: > > > >> Does anyone have any suggestions for connecting analog DID > > trunks? > > > > > > What is an analog DID trunk? > > > > You want to connect phones to your Asterisk? Connect to the PSTN? > > > >> I have some small locations that will have 2 analog DID trunks > > each, the only > >> solution that I can see will work will be using a channel > > bank and T1 card, > >> but it will be close to $1500 to terminate these DID > > trunks. Was hoping > >> someone had some experience using an ATA or TDM card and > > analog DID trunks. > >> > >> Rhino Channel Bank - $750 4 Port FXS module for channel bank - > >> $150 T1 Card - $500 > > > > > > This is for providing plenty of analog extensions (phones). Is that > > what you're after? > > > > -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED]<[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 > > > > !DSPAM:47b327c1163231152562594! > > > > -- > > > > > > ___ -- 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 > > > > !DSPAM:47b327c1163231152562594! > > - -- > James Finstrom > Rhino Equipment Corp. > Tel: 1-800-785-7073 ext. 6344 > FAX: +1 (480) 961-1826 > IP: asterisk.rhinoequipment.com ext 6344 > FWD: 633686 ext 6344 > > THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY > MATERIAL and is thus for use only by the intended recipient. If you > received > this in error, please contact the sender and delete the email and its > attachments from all computers. > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.6 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKB > Gxd6H7YOdzXfygVuBygzAw== > =51QY > -END PGP SIGNATURE- > > > ___ > -- 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] Microsoft Office Communications Server
We are interested in getting something working also, let me know how it goes. We are currently using LCS 2005 for IM, the only thing we want to add is the ability to update the "On the Phone" status in communicator. I have a test system on 1.6, but so far have been unable to update the presence information, would be interested if anyone has been able to do it will Office communications server. On Tue, Mar 11, 2008 at 10:37 AM, Razza <[EMAIL PROTECTED]> wrote: > On 10/03/2008, Matt Riddell <[EMAIL PROTECTED]> wrote: > > > > Has anyone done any integration with this? > > > > All I know so far is that it appears to use some non standard form of > > SIP. > > > > Any pointers? > > > ___ -- 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] DNS Changes never picked up with Asterisk 1.4.18 chan_sip?
Hello,
We're attempting to use Asterisk for distributing calls via SIP in a
large-scale speech recognition/VXML environment. We currently use DNS SRV
with weights and priorities to instruct VoIP gateways (not Asterisk) to
route calls to pools of servers. This works extremely well and provides for
load balancing, fail-over, and by setting the TTL low (several minutes) we
can easily take machines out of service for maintenance just by adjusting
the appropriate DNS SRV records. For business reasons, we now need to
achieve a similar setup using Asterisk instances hosted by a partner of
ours.
We've done a lot of reading on Asterisk and based on that our current
understanding is:
- chan_sip DOES now process DNS SRV weights and priorities (as of 1.4.X) but
only uses the first record (but correctly sorted by priority & weight)
- chan_sip DOES NOT use the dnsmgr for caching/update of DNS records
- Asterisk in general DOES NOT respect DNS TTL values (unclear if that was
addressed in recent versions)
In testing Asterisk 1.4.18, we are running into an even more basic issue --
changes to a host in DNS (with a single DNS SRV record) are never picked up
by Asterisk. Our simple test scenario is:
- Set SRV record in DNS for hostname X to point to server foo
- Place calls and observe that Asterisk correctly routes call to server foo
- Update SRV record in DNS for hostname X to point to server bar (with TTL
of hostname X originally set to 5 minutes)
- Subsequent calls still go to the original server foo, even after waiting
hours or even days
Asterisk never seems to pick up the changed DNS info.
There are vague references on the web to being able to configure Asterisk
with DNS related settings. ("Unless specifically configured, Asterisk
1.2does not honor DNS Time To Live (TTL)." from
*http://tinyurl.com/2ossv6) *We've tried doing:
[dnsmgr.conf]
enable=yes; enable creation of managed DNS lookups
refreshinterval=300 ; refresh managed DNS lookups every seconds
But this doesn't work, apparently since chan_sip doesn't use dnsmgr.
We figure that many other people must need to have Asterisk process updated
DNS info and that we must just be missing some configuration. Is that the
case or is this a fundamental limitation of Asterisk in its current form?
Thanks in advance for the help,
-Joe
P.S. I'm subscribed to the list in digest form so a copy to this e-mail on
replies would be most appreciated. Thanks!
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Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
I have not found this to be so. As an end user, I have had excellent support from Digium on TDM400p. They have been more responsive the several times I had to call. Even cross shipping replacement cards (CC, required, of course). Cannot fault their support at all. joe a. >>> On 2/16/2008 at 12:53 AM, <[EMAIL PROTECTED]> wrote: > You are kidding, right ??? > > A small user that just buys one card won't get a good support from > Digium. It'll be just a waste of time on the phone. > > Practically any manufacturer gives similar support including ssh'ing > in the users box. > > Right now they push the user to buy a 4 channel echo canceller which > you can get from Octasic for $40. The card with 4 ports is retail > around $640. > > You can get OpenVox or another brand TDM400P compatible for 1/3 of > that + $40 for echo canceller. Now that's a Digium high marigin right there > .. someone has to pay the CEO salary and the mortgage for a > new building :) > > cheers > > On 2/15/08, James Finstrom <[EMAIL PROTECTED]> wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> I would say email Kevin what he asked. The problem with switching to a >> clone company is you get what you pay for. Sticking with Digium you at >> least have support. and 3 clone cards and hours of troubleshooting >> later you will wish you hadn't been all cheap. > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP
Looks like it is part of the 1.6 Beta. >From the Change Log: 2008-01-18 22:04 + [r99080-99085] Russell Bryant <[EMAIL PROTECTED]> * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) On Feb 13, 2008 4:21 PM, Razza <[EMAIL PROTECTED]> wrote: > I am aware there is a SIP over TCP patch. Will this ever become part of > a release, if so are there any timelines? > 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
[asterisk-users] Analog DID
Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 ___ -- 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] ShoreTel <-> Asterisk Integration
Does anyone have experience using ShoreTel SIP trunks to integrate an Asterisk system? I am having trouble when the ShoreTel system transfers an incoming call from a SIP trunk to the voicemail system. From the SIP traffic, it looks like it negotiates a codec correctly, but once the RTP stream starts the call drops or there is no audio. I see errors in Asterisk such as: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 104 (Critical Request) Has anyone run into this before or have any ideas? Thanks, Joe ___ -- 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] ip phone suggestion for Asia?
I always forget about Linksys/Sipura phones. Yes they are very nice. I know there can be some contractual issues when you deal with Linksys/Cisco, but they are well constructed and I haven't heard anything bad. I do think you'd get better support with Snom/Grandstream, but this is coming from my experiences with Cisco more then Linksys. On Dec 20, 2007 3:24 PM, Daniel Cole <[EMAIL PROTECTED]> wrote: > We currently also use the Linksys SPA942 and SPA963 IP phones. They are > very nice phones, and very easy to manage. > > Cheers, > > Daniel Cole (CCNA) > Technical Support > > Ph: 1800 424 683 > Fax: 03 5221 7659 > e: [EMAIL PROTECTED] > w: hugonet.com.au > > > --- > > The information transmitted is the property of HugoNet and is intended > only for the person or entity to which it is addressed and may contain > confidential and/or privileged material. Statements and opinions expressed > in this e-mail may not represent those of the company. Any review, > retransmission, dissemination and other use of, or taking of any action in > reliance upon, this information by persons or entities other than the > intended recipient is prohibited. If you received this in error, please > contact the sender immediately and delete the material from any computer. > > P Please consider the environment before you print this e-mail or any > attachments. > > -Original Message- > From: [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] On Behalf Of Mindaugas Kezys > Sent: Friday, 21 December 2007 6:03 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] ip phone suggestion for Asia? > > Do not forget to evaluate Linksys SPA phones. Best I tried and not > expensive. > > We use them (SPA942) in our company. Everybody's happy. > > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > MOR - Advanced Billing for Asterisk PBX > > > -Original Message- > From: [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] On Behalf Of d tbsky > Sent: Thursday, December 20, 2007 6:34 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] ip phone suggestion for Asia? > > Hi: > i am surveying ip phones for our company. we will use them with > asterisk. > we have office in taiwan, hong kong,singapore and china. > cisco and polycom are too expensive for us. > we try several china brand ip phones. they are all cheap and some of > them have good quality. but most of them won't offer future firmware > support, which we think it's important for ip phones. > searching in the mail list, we found aastra is good, but they don't sale > to asia. grandstream looks good also.there are many grandstream users in > the list, can someone share any good or bad experience about grandstream > today? > if there are other good choice, please tell us!! > thanks a lot for your help!! > > Regards, > tbskyd > > ___ > --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] ip phone suggestion for Asia?
You get what you pay for. Snoms are good phones. Grandstreams are also good. I hear Snoms are easier to get around NAT and the seem like higher quality construction. Grandstreams are great for cheap and easy set-ups. I remember one guy telling me he buy up a case and if anything goes wrong with a unit he has a couple spares to go in its place. Over all... if you're looking for setting up office phones, Snom, Polycom, and Aastra look/feel nice. If you're looking to set up small offices or call centers on the cheap, Grandstreams are OK. I personally like Grandstream for home use, but I use Polycom for work, and I hear good things about Snom. On Dec 20, 2007 12:06 AM, d tbsky <[EMAIL PROTECTED]> wrote: > hi Joel: > > thanks a lot for your reply. i forgot snom :) > i wrote a snom employ found in this email list, but got no reply. > i saw there are huge complain about grandstream firmware this year. > grandstream seems response and solve some of them. i wonder if their > product is ok now. > i don't know the situation of snom, will they response to user's request? > thanks again for your kindly help!! > > Regards, > tbskyd > > > > 2007/12/20, Joel Hill <[EMAIL PROTECTED]>: > > Hi tbskyd, > > > > We have found that the Grandstream's are not that great a phone. One of > > our best sellers is the Snom range and I know that the Australian > > supplier spends half his time in Hong Kong so you shouldn't have any > > problems getting so over there. They are a little more expensive than > > the Grandstream's but cheaper than the Polycoms around that Aastra price > > range. > > > > Cheers, > > > > Joel. > > > > On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: > > > Hi: > > >i am surveying ip phones for our company. we will use them with > asterisk. > > >we have office in taiwan, hong kong,singapore and china. > > >cisco and polycom are too expensive for us. > > >we try several china brand ip phones. they are all cheap and > > > some of them have good quality. but most of them won't offer future > firmware > > > support, which we think it's important for ip phones. > > >searching in the mail list, we found aastra is good, but they don't > sale to > > > asia. grandstream looks good also.there are many grandstream users in > the list, > > > can someone share any good or bad experience about grandstream today? > > >if there are other good choice, please tell us!! > > >thanks a lot for your help!! > > > > > > Regards, > > > tbskyd > > > > > > ___ > > > --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] shoreline IP100 aka Polycom 500 boot problem
hold 468* or 135 when unit is booting... or not booting... hold those down when you first try to start the phone. then load the firmware/bootrom/etc On Dec 19, 2007 3:55 PM, Joe <[EMAIL PROTECTED]> wrote: > if that's all they do, then it doesn't sound good. > > There is a way to do a system format when they first load up, but the keys > you hold down escapes me at this moment and I seem to remember it took some > searching to find. I'll see if I can track it down. > > On Dec 19, 2007 11:56 AM, Robert Augustyn <[EMAIL PROTECTED]> > wrote: > > > Load the sip on it and you good to go ... assuming the phones are ok > > ... > > > > -- > > *From:* [EMAIL PROTECTED] [mailto: > > [EMAIL PROTECTED] *On Behalf Of *dave cantera > > *Sent:* Wednesday, December 19, 2007 12:27 PM > > *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > > Discussion > > *Subject:* [asterisk-users] shoreline IP100 aka Polycom 500 boot problem > > > > > > my client purchased a couple of shoreline ip-100 phones... I managed to > > get them to Not boot up... shows the polycom logo then goes blank... > > looks like the want mcgp... oh, mgcp... > > > > is there a solution for this? besides sending it back to polycom? > > daveC > > > > > > ___ > > --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] shoreline IP100 aka Polycom 500 boot problem
if that's all they do, then it doesn't sound good. There is a way to do a system format when they first load up, but the keys you hold down escapes me at this moment and I seem to remember it took some searching to find. I'll see if I can track it down. On Dec 19, 2007 11:56 AM, Robert Augustyn <[EMAIL PROTECTED]> wrote: > Load the sip on it and you good to go ... assuming the phones are ok ... > > -- > *From:* [EMAIL PROTECTED] [mailto: > [EMAIL PROTECTED] *On Behalf Of *dave cantera > *Sent:* Wednesday, December 19, 2007 12:27 PM > *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > Discussion > *Subject:* [asterisk-users] shoreline IP100 aka Polycom 500 boot problem > > > my client purchased a couple of shoreline ip-100 phones... I managed to > get them to Not boot up... shows the polycom logo then goes blank... > looks like the want mcgp... oh, mgcp... > > is there a solution for this? besides sending it back to polycom? > daveC > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP fails to register
Thanks, I believe that is what I was looking for. joe a. >>> On 12/14/2007 at 3:44 PM, "Zaheer K. Master" <[EMAIL PROTECTED]> wrote: > Hi Joe, > In your SIP.conf, under [general] try setting "externip=XXX.XXX.XXX.XXX" to > your public IP address. > > Hope this helps, > Zaheer > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Joe Acquisto > Sent: Friday, December 14, 2007 2:44 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP fails to register > > Trying to setup SIP to register with a VOIP provider. I am behind a > firewall (IPCOP) with NAT. > > Getting this, in CLI with SIP debug on. > > Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060: > REGISTER sip:voip-xxx.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport > From: ;tag=eufhksk > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 103 REGISTER > User-Agent: Asterisk PBX > Max-Forwards: 70 > Expires: 120 > Contact: > Event: registration > Content-Length: 0 > > I suspect there is something, somewhere, where I can tell it the "Contact" > should in fact be my public IP, not the local IP. > > Anyone know? Or know what else it might be? I am almost 100% certain my > credentials are correct. > > joe a. > > > ___ > --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] SIP fails to register
Trying to setup SIP to register with a VOIP provider. I am behind a firewall (IPCOP) with NAT. Getting this, in CLI with SIP debug on. Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060: REGISTER sip:voip-xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport From: ;tag=eufhksk To: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 I suspect there is something, somewhere, where I can tell it the "Contact" should in fact be my public IP, not the local IP. Anyone know? Or know what else it might be? I am almost 100% certain my credentials are correct. joe a. ___ --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] Fwd: re: Iax and ZAP
>>> On 12/12/2007 at 8:35 AM, Joe Acquisto wrote: I have a working system with two fxo and two fxs channels. I recenlty got > an > > IAX2 account I would like to use also. . . . > > . . . the outgoing calls attempt > > to go out over the ZAP channel. I can see this, via the CLI, with debugs > on. > > After correcting my errors, I can receive calls. Outgoing calls remain non functional, insisting on going out over the ZAP channel. I think I would like to maintain the ZAP channels and use iax on occasion. How about something like pressing *99 (or something), presenting a dial tone (or some noise) that would signal to enter the intended number? I am not getting how to "direct" which channel/trunk to use. joe a. ___ --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] Iax and ZAP
>>> On 12/11/2007 at 7:22 AM, Joe Acquisto wrote:
> I have a working system with two fxo and two fxs channels. I recenlty got an
> IAX2 account I would like to use also.
>
> While I have gotten the IAX2 channel to "register", it remains non
> functional, as the incoming calls, go nowhere and the outgoing calls attempt
> to go out over the ZAP channel. I can see this, via the CLI, with debugs on.
>
> I strongly suspect this is a dial plan/config problem with my setup, but I
> am currently at a loss. I've found no examples (that make sense) via google.
>
> On incoming calls, I get a "no such context/extension". I do not have trunk
> defined for IAX. Outgoing did not work when I did, either.
>
> joe a.
Managed to get calls to answer (no more "no such context/extension"), but they
just hangup immediately.
Below is a sanitized snippet of the debug output:
--
tel1*CLI>
-- Accepting AUTHENTICATED call from nn.nnn.nnn.nnn
> requested format = ulaw,
> requested prefs = (g729|ulaw|g726|gsm),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
-- Executing [EMAIL PROTECTED]:1] Answer("IAX2/iaxprovider-3", "") in new
stack
-- Executing [EMAIL PROTECTED]:2] Dial("IAX2/iaxprovider-3", "SIP/200|20")
in new stack
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT
Timestamp: 00070ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569]
FORMAT : 4
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER
Timestamp: 00073ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569]
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:3] Goto("IAX2/iaxprovider-3",
"aa_menu3|s|2") in new stack
-- Goto (aa_menu3,s,2)
-- Hungup 'IAX2/iaxprovider-3'
Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: HANGUP
Timestamp: 00084ms SCall: 3 DCall: 00247 [nn.nnn.nnn.nnn:4569]
CAUSE CODE : 3
--
A snippet of the dial plan.
--
[incoming-iaxprovider]
exten => xx,1,Answer()
exten => xx,2,Dial(SIP/200,20)
exten => xx,3,goto(aa_menu3,s,2)
exten => s,n, Hangup
-
joe a.
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[asterisk-users] Iax and ZAP
I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to "register", it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config problem with my setup, but I am currently at a loss. I've found no examples (that make sense) via google. On incoming calls, I get a "no such context/extension". I do not have trunk defined for IAX. Outgoing did not work when I did, either. joe a. ___ --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] Checking Dial Status
All,
I'm looking for a creative way to do this...
I've got a Polycom 650 as well as a standard cordless phone connected to
the Zap channel...
My Dialplan in extensions.conf looks like this:
exten => 9007,1,Dial(zap/1r4&SIP/polycom-9007,20,r)
exten => 9007,2,Answer
exten => 9007,3,GoTo(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(9007,u)
exten => s-BUSY,1,Voicemail(9007,b)
What I would like to have happen is that if I have the "Do Not Disturb"
option enabled on my Polycom 650, I would like the user to get the busy
voicemail greeting...
It works flawlessly if I do NOT dial the zap/1 channel... It also works
flawlessly if I enable "Do Not Disturb" on BOTH the SIP and Zap channels
being dialed.
Basically what I'd like to do is if the SIP channel is Busy, jump
immediately to busy voice mail greeting... But if the SIP channel is not
Busy, Dial both the SIP and ZAP channels...
Is there any way to do this correctly and probe my SIP phone before I dial
any channels?
Thanks
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Re: [asterisk-users] Polycom 601 stops ringing
>>> On 12/7/2007 at 2:33 PM, Doug <[EMAIL PROTECTED]> wrote: > At 10:58 12/7/2007, Joe Acquisto wrote: > >I have an odd issue, where a polycom 601 stops ringing, or more > >properly, maybe, stops *being* rung, when a call comes in. Other > >phones/extensions, continue to work fine, they being run at the same time. > > > >My dial plan works fine (?) seems it will ring properly, right after > >a reboot. It works fine for outgoing calls at all times. > > > >Hints? > > Is it behind a firewall? > > > > >joe a. > > My entire network is behind a firewall, but there is only a switch between asterisk and the phones. joe a. ___ --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] Polycom 601 stops ringing
I have an odd issue, where a polycom 601 stops ringing, or more properly, maybe, stops *being* rung, when a call comes in. Other phones/extensions, continue to work fine, they being run at the same time. My dial plan works fine (?) seems it will ring properly, right after a reboot. It works fine for outgoing calls at all times. Hints? joe a. ___ --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] Asterisk 1.4.15 sip.conf register
Hello, I recently upgraded from Asterisk 1.4.0 to 1.4.15... I am registering to a sip provider in my sip.conf as below [general] register=>user:password:[EMAIL PROTECTED]/extension Later down in my sip.conf I have the definition for that service provider as follows [serviceprovider] type=peer host=x.x.x.x port= outboundproxy=t.t.t.t . . . With asterisk 1.4 it would know to look in the peer definition for the IP address information... Now it appears that asterisk 1.4.15 is trying to do a DNS looking on "serviceprovider"... Of course that's coming back as an unknown host, and it no longer registers. Any steers as to how I can get asterisk 1.4.15 to look at the peer definition for the address info would be appreciated. Thanks Much! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip to ATA?
>>> On 11/27/2007 at 12:26 PM, Ira <[EMAIL PROTECTED]> wrote: > At 06:01 AM 11/27/2007, you wrote: > >>I am hesitant to believe that I can simply plug my TDM400P >>(2fxo/2fxs) into these (ATA ?) jacks and call it good. >> >>Any insight? Am I better off ignoring their phone offering and >>setting myself up with an IAX or SIP provider? (and surplus-ing the >>card). I would end up needing more than their single line offering >>with a second line at $30/month (USD). Seems that might make more sense > Thanks for both the replies. Hope springs eternal. joe a. ___ --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] Sip to ATA?
Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding how the phone service works. All I can get is I plug my existing phones and answering machines into the back of the "cable modem" and am good to go. I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good. Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that might make more sense joe a. ___ --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] DST
>>> On 11/1/2007 at 4:22 PM, Turbo Fredriksson <[EMAIL PROTECTED]> wrote: > Quoting "Joe Acquisto" <[EMAIL PROTECTED]>: > >> My thanks to all. Problem resolved with the assistance. > > Would be nice if you posted HOW it was fixed to... I have this exact > same problem at home, but the work phones displays time correctly... > Sorry, did not want to take up more list space. To quote/snip/paste, from a very recent post (BJ Weschke) (and archives, polycom, etc) -: *** If you've got the files centrally managed, you can update the correct tags in sip.cfg to correct the situation. These are the "correct" settings for regions affected by the new DST regs: tcpIpApp.sntp.daylightSavings.enable="1" tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" tcpIpApp.sntp.daylightSavings.start.month="3" tcpIpApp.sntp.daylightSavings.start.date="8" tcpIpApp.sntp.daylightSavings.start.time="2" tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" tcpIpApp.sntp.daylightSavings.stop.month="11" tcpIpApp.sntp.daylightSavings.stop.date="1" tcpIpApp.sntp.daylightSavings.stop.time="2" tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="0" *** joe a. ___ --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] DST
My thanks to all. Problem resolved with the assistance. joe a. >>> On 11/1/2007 at 1:43 PM, "Joe Acquisto" <[EMAIL PROTECTED]> wrote: > My Polycom phones are displaying time, off by one hour. Seems they are on > the old DST rules. How do I fix this? > > joe a. > > > ___ > --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] DST
My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a. ___ --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] FAX detection not working
>>> On 9/29/2007 at 3:27 PM, Lee Howard <[EMAIL PROTECTED]> wrote:
> Joe Acquisto wrote:
>
>>As I understand it, I must have faxdetect = incoming to enable detection of
> the fax tone.
>>Then, I must have a [fax] context to pickup the line and send it to whatever
> extension the FAX device is on.
>>
>
> It's a "fax" extension in the context where the call is at... not a fax
> context in the dialplan.
>
> Lee.
>
I don't follow. Sorry.
Now might be a good time to post this, since Tzafrir asked, it looks very much
like bits I have seen on the net. I did see what appeared to be the analog_fax
part when checking at CLI.
So, I would surmise it detected the FAX and is trying to deal with it, but the
number derived via LDAPget is hosed? It just ends up hanging up and not
dialing any extension.
{begin snippet]
[ext-fax]
exten => s,1,Answer
exten => s,2,Goto(in_fax|1)
exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax|1)
exten => in_fax,2,Macro(faxreceive)
exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf -
${FAXFILE}.pdf)
exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from
${CALLERID(num)} ${CALLERID(name)}" --attachment ${CALLERID(num)}.pdf --type
application/pdf --file ${FAXFILE}.pdf)
exten => in_fax,5,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten => in_fax,6,Hangup
exten => analog_fax,1,GotoIf($[foo${FAX_RX} = foo]?3:2)
exten => analog_fax,2,LDAPget(DIAL=DeviceDial/${FAX_RX})
exten => analog_fax,3,Dial(${DIAL}|20|d)
exten => analog_fax,4,Hangup
exten => out_fax,1,txfax(${TXFAX_NAME}|caller)
exten => out_fax,2,Hangup
exten => h,1,Hangup()
[end snippet]
joe a.
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