Re: [Asterisk-Users] installing Asterisk from source
Daniel Mikusa wrote: Look in the Makefile for the variables 'INSTALL_PREFIX' and 'PREFIX' they control where Asterisk is installed. Dan Jeremy Jones wrote: Is there a way to install Asterisk from source and not stomp on your already existing Asterisk installation? I don't see a configure script and it looks like it's trying to find stuff in /etc/asterisk and in /usr/lib/asterisk and probably other places. - Jeremy Jones ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks! That did it! - jmj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] installing Asterisk from source
Is there a way to install Asterisk from source and not stomp on your already existing Asterisk installation? I don't see a configure script and it looks like it's trying to find stuff in /etc/asterisk and in /usr/lib/asterisk and probably other places. - Jeremy Jones ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor() creating choppy audio files
I've created the following really simple dialplan:
;;;exten = 100,1,Monitor(gsm,ast_mon_${TIMESTAMP})
exten = 100,1,Monitor(WAV,ast_mon_${TIMESTAMP})
exten = 100,2,Dial(SIP/jmjones-l1,20,Ttr)
exten = 100,3,StopMonitor( )
exten = 100,4,VoiceMail(u100)
exten = 100,5,Hangup()
I've alternated between gsm, WAV and wav and have encountered the same
results: the live audio between the calling parties is goodgood
enough for what I want to accomplish, but the recorded audio quality is
a little choppy.
Basically, what I'm wanting is for one person to be able to call another
person via SIP and have asterisk handle the conversation so it can be
recorded. Both ends of the conversation are using gsm and as I
mentioned, I've tried recording to a gsm file as well as wav and WAV.
During the call, CPU isn't pegged, nor is disk IO, nor is network IO.
Has anyone seen anything like this? Anyone have any alternatives they'd
like to propose?
BTW, I'm running Asterisk 1.0.9 on Ubuntu Breezy and have run Linphone
on Linux, SJPhone on Linux, and SJPhone on Windows all with the same result.
Any help is greatly appreciated.
- Jeremy Jones
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Re: [Asterisk-Users] Sangoma VS. Digium
On Thu, 2005-03-31 at 02:43 -0500, Brian Capouch wrote: My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. Come on! If Digium started manufacturing tires, would i need to put 'em on my car to keep on the favorable side of karma? Digium makes telephony hardware that they sell on the open market. They also subsidize Asterisk development. If Digium finds itself unable to compete in an open market, do you believe Asterisk development would cease? Now, if another company makes hardware that they advertise as working with Asterisk, and that hardware is (hypothetically, of course, as I've never used Sangoma hardware) 1) better priced, and 2) functionally superior, does this not make Asterisk itself more accessible? As more integrators and developers are drawn to Asterisk, more and more subsidies will come to the development of Asterisk. By releasing the source to the community, Digium has ensured that the community can grow to the point where Asterisk development can sustain itself w/out worrying about the economic survival of any one particular business enterprise. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. If the point of buying Digium hardware is sentimental -- i.e. if you insist on paying Digium for your cards because you want to thank them for being nice and helping with Asterisk -- that's one thing (although, for all you know, there could be lots of really nice people working for Sangoma, too). If the assertion is that Asterisk will die w/out Digium, I, for one, think that's nonsense. Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small Local telco (wifi voip) some experiences with * ??
Paco Perez wrote: Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before I'd advise against it. I recently worked for a small telco/isp that tried this. I've been out of the company for a while now, but since I left, they've trashed the wireless gear replaced with fiber. From my experience, you'll get decent quality sometimes -- maybe even great quality most of the time -- but when little Timmy can't talk to Granny on Sunday afternoon 'cuz the Moon is in Virgo your wireless network's gone flaky, you'll hear all about it. Use a network you can trust as much as you expect to trust a pots line, or don't bother, I'd say. Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Ring Mystery
Hi David On Tue, 2005-01-25 at 09:10 -0800, David Shaw wrote: Out of the blue extension 100 will ring once. This will happen 3-4 times a day. I have checked the logs and no incoming calls. I have extension 100 and 101 going to a SPA-2000 ATA. Any ideas on this?? this is probably that little irritating feature of the sipuras -- 1/2 ring when you have voicemail. check the sipura configuration tell it not to ring on message waiting. jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zap, agents, ackcall
Hi, With ackcall=no set in agents.conf, agents on sip and iax channels are not required to press # to answer. However, this setting does not seem to do anything for agents on zap channels. This is with asterisk-1.0.2 and zaptel-1.0.2. I saw one post a while back about this, but no answer. Anyone have insight? -- Jeremy Jones [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ringing after hangup
Hello, I'm having a bit of trouble getting asterisk + tdm400p + rhino channel bank set up just right. The problem is a single ring on a telephone connected to the channel bank after the line is hung up. I've seen a couple messages similar to this, but I haven't seen any resolutions to the problem. Using the tdm400p card, with the first port connected to a 24-port fxs rhino channel bank; the second port connected to our provider's pri line, here's /etc/zaptel.conf: span=1,0,0,esf,b8zs span=2,1,0,esf,b8zs fxoks=1-24 bchan=36-47 dchan=48 loadzone=us defaultzone=us And /etc/asterisk/zapata.conf looks like this: [channels] group=1 context=default signalling=pri_cpe switchtype=ni1 echocancel=yes echocancelwhenbridged=yes callerid=asreceived echotraining=400 channel=36-47 context=system signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes callerid=asreceived echotraining=400 threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes ; callerid = 1 1 channel = 1 ; callerid = 2 2 channel = 2 ; .. and so on... I have tried switching the signaling from fxo_ks to fxo_ls. When I do that, there's no ringing after the phone is hung up, but there's no dtmf picked up by asterisk either... Anyone with ideas? Thanks, -- Jeremy Jones [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ringing after hangup
TYPO ALERT... On Fri, 2004-12-10 at 14:25 -0700, Jeremy Jones wrote: ..snip... Using the tdm400p card, with the first port connected to a 24-port fxs ..snip... That should read ...T400P card..., not ...tdm400p card... -- Jeremy Jones [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One More IP Phone for interoperability with Asterisk
Hi all, This is phone and the ATA is available soon from http://www.eezeephone.com priced at $75.00 each. Both have SIP+H323 and MGCP (also Net2Phone) compatibility. That site's a bit out of whack... Got to the voip phone product page at http://www.eezeephone.com/ezp_frm_productdetails.aspx?product_type=101 and try to match the pictures, descriptions, and prices. Looks like the matching-game homework my kindergartner brings home. If it weren't so dodgy, I might be interested in testing some of those 8x8 sip video phones. ...oooh, and my e-mail to their sales group was just returned as undeliverable -- mailbox full... jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Caller ID
Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display Asterisk as the cid name, nothing for number. Here's what I have in my mgcp.conf for the device: [2084728800103] host = dynamic context = westcomllc line = aaln/1 callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] When placing outbound calls (out our pstn gateway), I always replace cid name/number w/the main number name of the company, so that direction it's not an issue -- just internal calls. Anyone seen this have ideas about what to do with it? Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Caller ID
Hi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Cox Sent: Wednesday, July 28, 2004 7:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP Caller ID Try this: mgcp.conf [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 Aha! Yup, that did the trick. So order matters there... Thanks, jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Caller ID
Hi Duane (et alia), YES, because you could have an MGCP gateway device (more than one POTS line) ie. ours have 4 If so you would do something like this... [2084728800103] host = dynamic context = westcomllc callerid = Jeremy Jones 103 nat = no transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/1 callerid = Jeremy Jones #2 104 transfer = yes callwaiting = yes threewaycalling = yes cancallforward = yes mailbox = [EMAIL PROTECTED] line = aaln/2 ... etc... I have, actually, a gazillion 4-port mgcp devices from a (recently-obtained-by-8x8) company called Centile that I've _never_ been able to get to work properly w/* -- maybe this info'll help me here... ...snip... The fatality is that if asterisk is restarted, this database of mapping which was saved in memory; is now lost, so if a call came in and the end device was never rebooted/restarted (to accomidate the asterisk restart) the mapping did not exist, as it was not saved in a database and the call would fail. So I switched back to host=ip.ip.ip.ip Yeah, that's an issue here, too. We primarily have sip devices, though, at all our customer sites, so it's only a problem with _my_ phone internally, which so far doesn't bother me (I hate talking on the phone, anyway). If I just pick up the handset connected to the mgcp device hangup, that magic mapping is re-created. I'd love to be able to deploy some of these things, though, for our customers I really wouldn't like all the maintainance involved in setting up static dhcp assignments for all these mgcp devices tying addresses to each mgcp endpoint in mgcp.conf. We have, as I mentioned, tons of these mgcp thingies lying around waiting for use. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux sparc64 conferencing?
Ok, maybe a _wee_ bit esoteric, but... I've setup a developemnt system on a Sun Netra T1 running Aurora Linux (rh-7.3 for sparc64) -- just a few minor Makefile changes in codecs necessary. All in all, it's running nice. My only problem is lack of meetme or comprable features. I can't get zapdummy (these boxes use usb-ohci, not usb-uhci) to compile, and zaprtc sure won't work (no rtc module here...). Does anyone have any suggestions as to how I might either 1. weasle out of the timer requirements for meetme Or 2. use some as-yet-undocumented 3rd party conferencing plugin Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_oh323
Hi, Hello, I have been trying for a while to make the oh323 channel working but i didnt manage, i have everything compiled correctly but asterisk find somethign like an undefined symbol when it loads the oh323 module... Put the path to your openh323 libraries in your LD_LIBRARY_PATH environmental variable. You can put export LD_LIBRARY_PATH=/path/to/openh323/libs in your asterisk startup script. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] forced ring on dial?
I'd be willing to bet you have r in your dialout string (i.e.
something like: Dial(${TRUNK}/${EXTEN},120,r)...
Get rid of that in the outbound dialing, and you otta be ok.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Komito
Sent: Friday, June 25, 2004 8:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] forced ring on dial?
I am routing outgoing calls through a sip gateway. The calls
go through
no problem, however the ringing in the callers ear begins as
soon as the
last digit is dialed. This has two nasty side effects.
First, the caller
hears 1-2 more rings than the callee. Second, and more
importantly, if
the callee's line is busy, the caller continues to get hear
ringing, even
though the gateway has returned a busy indication.
The whole problem seems to be * is not waiting for the proper call
progress signal from the sip gateway before giving the caller a ring
indication. Is there any way to control this so that * waits for call
progress from the gateway before giving the caller the appropriate
indication, i.e., ring or busy tone? I have been told this
is a result of
setting * to forced ring and this should be turned off, but
of course,
on * it is probably called something else.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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RE: [Asterisk-Users] SS7 to Pri
Didn't I hear a week or two ago (on this list) that someone had taken it upon themselves to write an asterisk module for the openss7-modified digium t1/e1 cards? Maybe soon asterisk'll do it. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, June 25, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 to Pri On Fri, 2004-06-25 at 09:24, Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? I think it's called an ILEC or CLEC. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *69
Hello, I've managed to build in the last number repeat outlined at http://www.voip-info.org/wiki-Asterisk+last+number+repeat to call back the last person _I_ called from a particular phone, and now I'd like to try to do something similar for the common *69 -- call back the last number that called me. I assume I'll do part of this in my standard extension macro -- capturing the last callerid number that called a particular extension, along w/some sort of dbput command. Then in my [apps], a *69 extension that grabs that info dials it. Anyone have an example of such a configuration, or hints (i.e. what variable I'll catch in my standard extension macro)? Thanks! Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] *69
Got it!
How about just before you dial an extension you do a:
DBput(LAST/${EXTEN}=${CALLERID_NUMBER})
and then *69 does a:
DBget(dialNum=LAST/${CALLERID_NUMBER})
(from your extension) and you can dial it?
Here's what I've done--
In my extension macro:
exten=s,1,DBput(LastCIDNum/${DNID}=${CALLERIDNUM}) ; grab
CALLERIDNUM store it for the dialed number as LastCIDNum
exten=etc...
And in my [apps]:
; Return last call
exten = *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM}); read db value
LastCIDNum for this CALLERIDNUM (i.e. the extension making the call)
exten = *69,2,GotoIf($[${temp:0:3} = 208]?3:4) ; if it's area
code 208 (my local area), go to priority 3, otherwise priority 4
exten = *69,3,Macro(dialout,${temp:3}) ; call my
dialout macro, stripping the 208
exten = *69,4,Macro(dialout,${temp}) ; call my
dialout macro, using full string
exten = *69,103,Congestion ; no
key? congestion
exten = *69,104,Congestion ; no
key? congestion
Works like a friggin charm!
Jeremy Jones
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RE: [Asterisk-Users] *69
This:
; Return last call
exten = *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM}) ; read db value
LastCIDNum for this CALLERIDNUM (i.e. the extension making the call)
exten = *69,2,GotoIf($[${temp:0:3} = 208]?3:4)
; if it's area
code 208 (my local area), go to priority 3, otherwise priority 4
exten = *69,3,Macro(dialout,${temp:3})
; call my
dialout macro, stripping the 208
exten = *69,4,Macro(dialout,${temp}) ; call my
dialout macro, using full string
exten = *69,103,Congestion ; no
key? congestion
exten = *69,104,Congestion ; no
key? congestion
Should be this:
exten = *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM})
exten = *69,2,GotoIf($[${temp:0:3} = 208]?3:4)
exten = *69,3,Macro(dialout,${temp:3})
exten = *69,4,Macro(dialout,${temp})
exten = *69,102,Congestion
jeremy
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RE: [Asterisk-Users] *69
Andrew Thompson wrote:
Are your internal-to-internal calls handled seperately from
external-to-internal calls?
I see you are accounting for local versus ld calls, but what
about when the
person in the next cube over calls you?
Well, I have a dual-purpose asterisk setup -- hosted business pbx
services and residential services.
My dialout macro for residential services does this:
[macro-resdialout]
exten=s,1,DBput(RepeatDial/${CALLERIDNUM}=${ARG1})
exten=s,2,Dial(SIP/208${ARG1},120,rtT)
exten=s,3,Dial(${TRUNK}/9${ARG1},120,rtT)
exten=s,4,Busy
It first does some work for repeat dialing (not the *69 callback stuff),
moves on tries to get a local sip client, then goes on to dial out the
trunk if there's no local sip client.
So far, I've not yet worked out the business version...
jeremy
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RE: [Asterisk-Users] anyone use mailboxexists?
Hi all, Odd... I did a make update and how the MailboxExists works fine. However, it works just as the docs say: add 101 to priority if the box *does* exist, add 1 if not. I have tested it and this seems to be how it works. You may wish to test your flow and make sure it works as you think it does. Well, the cvs version I have (head from 3:00 mst june 14th) works as I expect with this macro... Let me triple-check... Here's one w/out voicemail: *CLI Check for res for is not a local user build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone -- Executing SetVar(SIP/my-pstn-gw-0813b410, NEW_MACRO_CONTEXT=residential-swidaho) in new stack -- Executing Macro(SIP/my-pstn-gw-0813b410, stdexten|2082874700|SIP/2082874700|1) in new stack -- Executing MailboxExists(SIP/my-pstn-gw-0813b410, [EMAIL PROTECTED]) in new stack -- Executing DBget(SIP/my-pstn-gw-0813b410, temp=CFIM/2082874700) in new stack -- DBget: varname=temp, family=CFIM, key=2082874700 Unable to find key '2082874700' in family 'CFIM' -- DBget: Value not found in database. -- Executing Goto(SIP/my-pstn-gw-0813b410, s|104) in new stack -- Goto (macro-stdexten,s,104) -- Executing Dial(SIP/my-pstn-gw-0813b410, SIP/2082874700|120|rtT) in new stack SIMPLE DIAL (NO URL) Setting NAT on RTP to -1 Unable to create channel of type 'SIP' == Everyone is busy at this time -- Executing Busy(SIP/my-pstn-gw-0813b410, ) in new stack == Spawn extension (macro-stdexten, s, 205) exited non-zero on 'SIP/my-pstn-gw-0813b410' in macro 'stdexten' == Spawn extension (did, 2082874700, 2) exited non-zero on 'SIP/my-pstn-gw-0813b410' update_user_counter() - decrement inUse counter is not a local user Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Not Found *CLI See where it goes to 104? That's where we end up if: no mailbox, no CFIM. When we find that the called device is unavail, it goes to 205 (I get busy signal). Now again w/voicemail: *CLI Check for res for is not a local user build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone -- Executing SetVar(SIP/my-pstn-gw-0813b830, NEW_MACRO_CONTEXT=residential-swidaho) in new stack -- Executing Macro(SIP/my-pstn-gw-0813b830, stdexten|2082874700|SIP/2082874700|1) in new stack -- Executing MailboxExists(SIP/my-pstn-gw-0813b830, [EMAIL PROTECTED]) in new stack -- Executing NoOp(SIP/my-pstn-gw-0813b830, ) in new stack -- Executing NoOp(SIP/my-pstn-gw-0813b830, ) in new stack -- Executing NoOp(SIP/my-pstn-gw-0813b830, ) in new stack -- Executing NoOp(SIP/my-pstn-gw-0813b830, ) in new stack -- Executing DBget(SIP/my-pstn-gw-0813b830, temp=CFIM/2082874700) in new stack -- DBget: varname=temp, family=CFIM, key=2082874700 Unable to find key '2082874700' in family 'CFIM' -- DBget: Value not found in database. -- Executing Goto(SIP/my-pstn-gw-0813b830, s|9) in new stack -- Goto (macro-stdexten,s,9) -- Executing Dial(SIP/my-pstn-gw-0813b830, SIP/2082874700|25|rtT) in new stack SIMPLE DIAL (NO URL) Setting NAT on RTP to -1 Unable to create channel of type 'SIP' == Everyone is busy at this time -- Executing VoiceMail(SIP/my-pstn-gw-0813b830, [EMAIL PROTECTED]) in new stack voicemail/residential-swidaho/2874700/unavail doesn't exist, doing what we can Ooh, format changed from UNKN to ULAW -- Playing 'vm-theperson' (language 'en') Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found -- Playing 'digits/2' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (macro-stdexten, s, 110) exited non-zero on 'SIP/my-pstn-gw-0813b830' in macro 'stdexten' == Spawn extension (did, 2082874700, 2) exited non-zero on 'SIP/my-pstn-gw-0813b830' update_user_counter() - decrement inUse counter is not a local user *CLI So here it hits 2, 3, 4, 5 (all the noops) and on down to 6, as I expected it would when I *do* have a vm box, but the opposite of what the docs say. I suppose I may need to re-write this macro after I update, eh? Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Directory dial by name
If the user to whom that vm is assigned goes through the setup process records their name, that is played. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Harold Workman Sent: Monday, June 21, 2004 11:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Directory dial by name Just a quick question. I setup Directory dial by name, and I read it looks at the Voicemail config to determine who you want to connect to. The thing I dont like is when it finds a match it reads the extension instead of their name. Is there a way to have it read the name in the voicemail config rather than the extension? Thanks, Harold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] anyone use mailboxexists?
Michael,
From the docs, it looks like MailboxExists() will add 101 to the
priority if the box *does* exist and goes to the next priority if not.
I think the show application mailboxexists documentation is wrong. I
believe it's the other way around. It does exits? Jump to next
priority. It doesn't? Jump to n+101. Here's my extension macro (sift
out the forwarding stuff if you don't like that), and it works:
[macro-stdexten]
exten=s,1,MailboxExists(${MACRO_EXTEN:[EMAIL PROTECTED]);If
mailbox exists continue at 2, otherwise goto 102
exten=s,2,NoOp ;Filler
exten=s,3,NoOp ;Filler
exten=s,4,NoOp ;Filler
exten=s,5,NoOp ;Filler
exten=s,6,DBget(temp=CFIM/${ARG1}) ;Get
CFIM key, if not existing, goto 107
exten=s,7,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,8,NoOp ;Filler
exten=s,9,Dial(${ARG2},25,rtT) ;Dial device for
25 seconds, goto 10 if busy, goto 110 if unavailable
exten=s,10,NoOp ;Filler
exten=s,11,DBget(temp=CFBS/${ARG1}) ;Get
CFBS key, if not existing, goto 112
exten=s,12,Dial(${TRUNK}/9${temp}) ;Forward
on busy or unavailable
exten=s,102,DBget(temp=CFIM/${ARG1});Get CFIM key,
if not existing, goto 203
exten=s,103,Dial(${TRUNK}/9${temp})
;Unconditional forward
exten=s,104,Dial(${ARG2},120,rtT) ;Dial
device for 120 seconds, goto 105 if busy, goto 205 if unavailable
exten=s,105,DBget(temp=CFBS/${ARG1});Get CFBS key,
if not existing, goto 206
exten=s,106,Dial(${TRUNK}/9${temp}) ;Forward
on busy or unavailable
exten=s,107,Goto(s,9) ;Goto 9
exten=s,110,Voicemail(u${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if unavailable
exten=s,111,Hangup ;Hang up
the channel when vm exits
exten=s,112,Voicemail(b${MACRO_EXTEN:[EMAIL PROTECTED]) ;To VM
if busy
exten=s,113,Hangup ;Hang up
the channel when vm exits
exten=s,203,Goto(s,104) ;Goto
104 for accounts w/out vm
exten=s,205,Busy() ;Busy
signal if busy no vm
exten=s,206,Busy() ;Busy
signal if no answer in 2 min no vm
It's a little ugly w/all those NoOps, but I think I need those to get
the priorities right.
Jeremy Jones
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RE: [Asterisk-Users] LDAP synchronization script
David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HOW-TO DIFF
Type: user$ man patch -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Devine Sent: Tuesday, June 08, 2004 11:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] HOW-TO DIFF How do I patch an * file if all I have is the .diff file? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] videosupport = yes -- how to use it?
I can't speak for general cases, but I know when I've tried to set videosupport=yes, my as5300 can no longer speak w/*. I wonder if it can be set per peer - haven't tried that... jeremy jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Mielke Sent: Monday, June 07, 2004 9:10 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] videosupport = yes -- how to use it? Hi all, can Asterisk be used as a videoconference server or the like when enabling 'videosupport=yes' ? if so, how do I use it? is there any recommended SIP/Video-client for both Windows and Linux? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] parking in multiple contexts
Hi all, If I have multiple extension contexts for different businesses sharing an * pbx, and I include = parkedcalls in each of these separate contexts, would someone at biz-1 be able to pick up a parked call that someone in biz-2 parked? I _don't_ want this to happen -- I'd like to be able to include a different parkedcalls context in each business' context, and prevent someone at, say, the mortgage broker's office from picking up a parked call from the collection agency's context. Anyone have an idea of how to implement this? Thanks Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Receptionist manager program.
I think what he means is this: I can have extension 104 defined in multiple contexts, for instance if I host virtual pbxs for multiple customers on one * box. The syntax of my * conf files requires exten@context if I want to differentiate between these extensions. If you're using the * box for one business, and you ensure that the same extension is not used in multiple contexts you're ok. But otherwise... Same goes for voicemail -- I can have voicemail box 104 in the voicemail context [great_customer], and another 104 in voicemail context [really_great_customer]. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Thursday, June 03, 2004 3:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. This is MY prefs.conf Serverip,192.168.1.40 Port,5038 UID,mark PWD,mysecret MyExt,104 ServerIp = Asterisk Server Port = Port for Manager UID = Manager User PWD = Manager Password MyExt = Extention for Asterisk Manager to Monitor, Transfers, Check VoiceMail etc (my Extention is 104) MyExt is so the person running the application CANT transfer someone elses calls accidentally. It will only effect ext 104's calls. Did I cearify your question? Kyle Brett Nemeroff wrote: Ok, Maybe I'm missing something here.. What does Extension mean without Context also being defined. I don't know what to set in my prefs.conf file for extension... ?? -Brett Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan [EMAIL PROTECTED] Organization: Nuvo Technologies To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. Reply-To: [EMAIL PROTECTED] I put a new version up last night. Caller ID shows up on the buttons. This time IAX is fixed. Works at home and at work through FWD. http://www.easyhomenetworks.com/AstRec/ Has anyone had anyother bugs popup other than the IAX problem? Some people are asking why the screen shot has more buttons than the alpha version. We are going to get the bugs worked out of the existing buttons before we add more features. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfering with Grandstream Phones
The gs handytone 286 user manual at the gs website lists call transfer/call forward as not yet implemented (the firmware version listed on the manual is out of date, thoughg). I sent a query to gs tech support a month or two ago and received this: Call waiting and Caller-ID are implemented in HT-286. Call transfer is implemented but not activated yet. We expect to release a firmware to enable Call transfer in about a week. Call forward is not supported yet. I haven't tried any newer firmware, though. jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Saturday, May 08, 2004 2:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Transfering with Grandstream Phones On Sat, 2004-05-08 at 20:43, Ryan Courtnage wrote: On 8-May-04, at 12:09 PM, Paul Tyreman wrote: I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have the same problem (attempting to transfer a call made to my BT102 will result in that call being disconnected/hung). Workaround is to use '#' to transfer instead of the 'transfer' button on the phone. I also agree.. Using the '#' key is the only way to transfer. I'm running Software Version: 1.0.4.63 Nothing in the html menu mentions how 'transfer' might work - perhaps its a blank key waiting to be programmed one day??? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfering with Grandstream Phones
I'm talking about the ht, but I assume (big assumption, I know) that the sip stack is the same between the two. jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Monday, May 10, 2004 9:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Transfering with Grandstream Phones On Mon, 2004-05-10 at 09:12 -0600, Jeremy Jones wrote: The gs handytone 286 user manual at the gs website lists call transfer/call forward as not yet implemented (the firmware version listed on the manual is out of date, thoughg). I sent a query to gs tech support a month or two ago and received this: Call waiting and Caller-ID are implemented in HT-286. Call transfer is implemented but not activated yet. We expect to release a firmware to enable Call transfer in about a week. Call forward is not supported yet. I haven't tried any newer firmware, though. Are you talking about the Grandstream HT286 or the Grandstream BT101? The original post was talking about the BTs. In both cases I can transfer with no problems using 1.0.4.63 firmware on both units. In the case of the BT using the Transfer button and in the case of the HT using the R button on the Siemens Gigaset. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS load-balancing SRV records
Let's say I have a third-party device acting as a sip--pstn gateway, a cluster of three asterisk servers, and a teensy bit of dns knowledge. Let's now say those asterisk servers are a1.company.com at 192.168.0.1, a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3. 1. If I setup round-robin dns like so: asterisk.company.com. IN A 192.168.0.1 asterisk.company.com. IN A 192.168.0.2 asterisk.company.com. IN A 192.168.0.3 2. and normal A records for the servers like this: a1.company.com. IN A 192.168.0.1 a2.company.com. IN A 192.168.0.2 a3.company.com. IN A 192.168.0.3 3. and srv records like so: _sip._udp.company.com IN SRV 20 0 5060 a1.company.com _sip._udp.company.com IN SRV 30 0 5060 a2.company.com _sip._udp.company.com IN SRV 40 0 5060 a2.company.com 4. and configure all my sip clinets to register to asterisk.company.com 5. and tell the 3rd-party sip--pstn gateway to use srv records that all inbound calls from the pstn should go to the realm company.com 6. and all the configs on the asterisk servers are identical... would I have successfully setup load-balancing across my asterisk servers? Thanks, Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 729 licence on scsi
You need to mount a cd before running the Registration binary. Also, you'll need to mount a cd any time you want to start/restart asterisk (as I discovered this morning...) jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, May 07, 2004 2:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 729 licence on scsi I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? Who do I talk to now? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uClibc patch?
Hi, I've been searching on an error I'm getting trying to compile against uClibc, related to enum support. I found reference in an earlier thread (http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html) to a patch adding an Makefile option to remove enum support. Anyone have that diff file lying around? Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls to Cisco PSTN gateway
Make sure you don't have videosupport=yes in sip.conf when using as5300. I found mine doesn't like that much got that codec error. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Radius Sent: Thursday, April 15, 2004 2:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Calls to Cisco PSTN gateway Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial(SIP/ata186-c1cf, SIP/[EMAIL PROTECTED]:5060|30|r) in new stack -- Called [EMAIL PROTECTED]:5060 mailto:[EMAIL PROTECTED]:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp: Error in codec string 'ideo 0 ' Asterisk was configured with allow=ulaw. Any idea for this problem?? Thanks. Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Nice elegant! Looks great. jj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: Thursday, April 01, 2004 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ANNOUNCE: Flash Operator Panel http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so expect some bugs. I appreciate any feedback you can give me. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer driving me batty
Can anyone help me get call transfers working?
I have grandstream handytone-286 sip ATAs. Attached to these, I have
Teledex B150D telephones. Are there magic lines I need in my sip peers
to enable these folks to transfer? A call rings in at, say, 7145551212,
goes to x100, and they want x101.
In extensions.conf, I have something like this:
[macro-bizstdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto
102
exten=s,2,Dial(${SIPTRUNK}/9${temp}) ; Unconditional forward
exten=s,3,Dial(${ARG2},20,rtT) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing,
goto 105
exten=s,5,Dial(${SIPTRUNK}/9${temp}) ; Forward on busy or unavailable
; No CFIM key
exten=s,102,Goto(s,3)
; No CFBS key - voicemail ?
exten=s,105,Voicemail([EMAIL PROTECTED])
exten=s,106,Hangup
exten=s,107,Voicemail([EMAIL PROTECTED])
exten=s,108,Hangup
[some-biz]
include = biz-outbound
include = 9208
include = app-dnd
include = app-callforward
exten = 555,1,Wait,2
exten = 555,2,VoicemailMain
exten = 555,3,Hangup
exten = 100,1,Macro(bizstdexten,100,SIP/7145551212100)
exten = 101,1,Macro(bizstdexten,101,SIP/7145551212101)
exten = 102,1,Macro(bizstdexten,102,SIP/7145551212102)
exten = 103,1,Macro(bizstdexten,103,SIP/7145551212103)
exten = 104,1,Macro(bizstdexten,104,SIP/7145551212104)
exten = 105,1,Macro(bizstdexten,105,SIP/7145551212105)
exten = 106,1,Macro(bizstdexten,106,SIP/7145551212106)
exten = 107,1,Macro(bizstdexten,107,SIP/7145551212107)
exten = 108,1,Macro(bizstdexten,108,SIP/7145551212108)
exten = 7145551212,1,Goto(100,1)
exten = 7145551213,1,Goto(100,1)
*
Here's a bit of sip.conf:
*
[7145551212100]
type=friend
username=7145551212100
secret=top_secret_word
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
context=some-biz
[EMAIL PROTECTED]
callerid=7145551212
[7145551212101]
type=friend
username=7145551212101
secret=top_secret_word
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
context=some-biz
[EMAIL PROTECTED]
callerid=7145551212
**
Anyone know what I otta be doing differently? I've told the ata's to do
dtmf via RTP (RFC2833). Should I change that to in-audio?
Thanks for any guidance,
Jeremy Jones
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[Asterisk-Users] A tidbit about one-way audio ethernet aliases
Hey all, Thought I'd share a curiosity I found when trying to use heartbeat software for asterisk failover (this may already be common knowledge to some/many, but I hadn't seen mention of it yet). The default ha-linux ip-takeover script uses ifconfig to create an ethernet alias to which a secondary IP address is assigned (i.e. eth0 is your main interface at 10.1.1.1, and the heartbeat script creates eth0:0 at 10.1.1.2). I had been testing my asterisk configuration w/out heartbeat 'til I thought it stable enough for production, then I turned on the heartbeat left the office to set up my first subscriber. Imagine my shame... No audio from pstn to subscriber (using sip ata behind nat). Seems the rtp stream doesn't appreciate being directed at a secondary address. So, swapping out the default ha-linux ip-takeover script for one that uses ip from the iproute2 package solved my problem. (Perhaps this is what Doichin Dokov had going on late last week?) Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer?
Ok I give up!
What do I have to do to my extensions to implement transfer? I've seen
mention of tacking a t or a T on the end of the dial string... So
here's a macro that I use for extensions:
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto
102
exten=s,2,Dial(${SIPTRUNK}/${EXTEN}) ; Unconditional forward
exten=s,3,Dial(${ARG2},20,T) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing,
goto 105
exten=s,5,Dial(${SIPTRUNK}/${EXTEN}) ; Forward on busy or unavailable
; No CFIM key
exten=s,102,Goto(s,3)
; No CFBS key - voicemail ?
exten=s,105,Voicemail([EMAIL PROTECTED])
exten=s,106,Hangup
exten=s,107,Voicemail([EMAIL PROTECTED])
exten=s,108,Hangup
Is this all wrong? Please tell me where... I have sip-speaking
Grandstream HandyTone-286 ATAs. What's the standard transfer sequence?
Flash-5number or something like that?
Thanks,
Jeremy Jones
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RE: [Asterisk-Users] Cisco AS5350 + Asterisk Configuration
Doichin, Here's a dial-peer from an AS5300 at 10.0.0.1: dial-peer voice 1 voip destination-pattern 714555 session protocol sipv2 session target ipv4:10.0.0.2 codec g711ulaw And a sip peer in * sip.conf at 10.0.0.2: [as5300] type=friend host=10.0.0.1 callerid=asreceived disallow=all allow=ulaw So far, this is a very straightforward setup - nothing fancy, just works. This particular AS5300 also speaks h323 to another voip soft-switch, so I have other dial-peers that match 714554, etc. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of NetOne Admin Sent: Wednesday, March 17, 2004 6:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5350 + Asterisk Configuration Hello, everybody! I need help connecting my Cisco AS5350 to *. What i want to do is forward all incoming calls coming from the E1 connected to the AS5350, to my * server, using SIP. How could this be done? Greetings, Doichin Dokov NetOne - Silistra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP native bridge vs. SIP reinvite
Hi,
I'm trying to get rtp media streams to run between endpoints rather than
through my * server, and I think I'm getting something wrong. I have an
AS5300 speaking both h323 (for a different voip system I run) and sip
for *. Dial-peers on the as5300 differentiate inbound from pstn to
different chunks of DID numbers between h323 and sip. I'm testing with
xlite on a PC.
So here's what I have:
Outbound trunks are defined in my extensions.conf that send _9whatever
to SIP/pstn_gw/${EXTEN}.
In sip.conf I have two friends, one for my xlite softphone, one for
pstn_gw:
[2085551212]
type=friend
username=2085551212
secret=1234
host=dynamic
canreinvite=yes
disallow=all
allow=ulaw
context=testme
mailbox=5551212
callerid=Jeremy Jones 2085551212
[pstn_gw]
type=friend
username=pstn_gw
disallow=all
allow=ulaw
context=default
canreinvite=yes
host=10.0.0.201
I can place a call from the PSTN to 5551212 successfully, and I can
place calls from xlite to the PSTN successfully. But in either case I
always see two sip channels active on *, and the endpoints (as5300
xlite) are sending their rtp via *. Here's what I see when I place a
call from xlite to:
*CLI -- Executing Prefix(SIP/2085551212-f04d, 9) in new stack
-- Prepended prefix, new extension is 93532533
-- Executing Dial(SIP/20825551212-f04d, SIP/pstn_gw/93532533) in
new stack
-- Called pstn_gw/93532533
-- SIP/pstn_gw-85a0 is making progress passing it to
SIP/2085551212-f04d
-- SIP/pstn_gw-85a0 answered SIP/2085551212-f04d
-- Attempting native bridge of SIP/2085551212-f04d and
SIP/pstn_gw-85a0
*CLI
*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
Format
10.0.0.201 9353253302e2e09e167 00103/00651 0ms ms
ULAW
10.0.0.100 2082874602E0541F6D-81 00102/03763 0ms ms
ULAW
2 active SIP channel(s)
*CLI
(I have a Prefix rule for outbound 'cuz this is a system for residential
users, and the as5300 has dial-peers that need a 9 prefix...)
The output in * is similar for inbound from PSTN to xlite.
I can send output from sip debug if that'd help.
Thanks,
Jeremy Jones
Network Nerd
WestCom, LLC
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[Asterisk-Users] DTA-310 Outbound Dialing
Hi folks, OK... I've successfully managed to get a DTA-310 from 8x8 to take inbound calls from the PSTN, into an AS5300, through asterisk. I can call the number associated from out there in the world, it rings, I can speak to the person on the other end just fine. However, I cannot seem to figure out how to place a call outbound through it. I do have outbound calling working fine with another SIP client, X-lite, so I'm pretty sure it's not my * setup -- I'm willing to share configs, though, if anyone thinks that'd help. This DTA-310 runs firmware version DTA version 1.0 US (8x8 00) as shown on the web interface homepage, which is the only version of the firmware (that I'm aware of) that allows tinkering with the SIP settings. I _think_ my problem has to do with the Dial Plan settings on the SIP configuration page. Anyone familiar with these things? By default, the dial plan setting reads: 1xx|x.T. Using this, or anything else I've experimented for that matter, I pick up the handset, dial some numbers, watch the asterisk console, and hear/see nothing. Dead air on the handset, no scrolling gibberish on the * console. Anyone have ideas? Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTA-310 Outbound Dialing
When I speak of the dial plan here, I'm referring to a portion on the
DTA-310 web pages, not my * dial plan. I've seen a couple posts about
setups like this:
* w/tdm card -- dta-310 -- packet8 network -- pstn
I'm not using the packet8 service, just the gear. Like this:
dta-310 -- * -- as5300 -- pstn
Dialing in works (i.e. pstn to dta-310), but I can't figure out the
dta-310 to pstn thing.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Sent: Thursday, February 26, 2004 1:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DTA-310 Outbound Dialing
On Thu, 26 Feb 2004, Jeremy Jones wrote:
I _think_ my problem has to do with the Dial Plan settings on the
SIP
configuration page. Anyone familiar with these things? By default,
the
dial plan setting reads: 1xx|x.T.
This is my dialplan for Packet8 / 8x8:
exten = _91[2-7]XXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91[2-7]XXNXX,2,Congestion
Dial 9 to get a line, only will accept calls non-toll free numbers
(toll
free calls are route elsewhere), dial the trunk card and the number,
strip the initial 9.
This recipe is locate under extensions.conf
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RE: [Asterisk-Users] The Smallest Asterisk Server Ever?
If asterisk'll compile against uclibc, it'll go on the toaster. Most toasters (and coffee grinders such) don't have enough flash memory for a full glibc... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion Sent: Tuesday, February 03, 2004 10:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? Greg Boehnlein wrote: Is this the smallest Asterisk server ever? :) WHY??? just kidding. That's pretty cool. Maybe if you kicked it up to 64 MB, you could create a 4-port sip fxo device and stop all of these posts about not being able to find one... This could be good news for the embedded front. Now, here's the real question: can you install it on a toaster? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?
Generally speaking, unless you're using an rtp proxy, the rtp audio should go client--client. H323 does the call setup and teardown and such, but the audio stream is usually direct. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Fargas Sent: Monday, February 02, 2004 4:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
Hi, Yep, I got the latest firmware (and the next-to-latest, and the next-to-next-to-latest, and one earlier yet) for SIP. The first three (firmware versions 1228, 1227, and 1226) all have that password protected Advanced Configuration page. The fourth one I found (version ) is a bit more open. It appears mum's the word on that password, none of the requests others have made in, say, vonage forums for that top secret password have met with much success. I suppose I could try to brute force it, but I'm fairly lazy, I'd just as soon wait for someone to just blurt it out on accident in casual conversation. I did poke around in the binary files found the html stuff, as well as what appear to be clear-text default options for stuff one would find on the protected Advanced Configuration page. It may be possible for me to use one of the newer firmware versions, changing the options in the firmware binary before sticking it on the device, but I imagine I'd screw up a checksum somewhere if I edit the file directly. Haven't tried yet. So, next stop: sip with the firware. Thanks, Jeremy You can get the latest SIP firmware from Packet8's TFTP server at 4.42.235.170 file name current. Read more about it here http://web.packet8.net/download/ Only problems is, in this version the advanced configuration page with the SIP setup is password protected. If you look at the downloaded file, you can see all the HTML stuff for the configuration pages. It may be possible to figure out or remove the password protection. The other option is to load an older version of the SIP firmware in which the SIP page is not protected. I'm sure someone has a copy of it. By the way, do you have a copy of the MGCP firmware in case you want to go back to it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
This one does mgcp... It's been used in conjunction with a hosted pbx system called Centile that 8x8 now owns. If there's a firmware image anyone knows of to make these do sip, I'd rather do that. But for now, mgcp help is what I need. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, January 23, 2004 9:08 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway The Packet8 8x8 DTA-310 that I have ran SIP when I was using it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway
Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line = aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/[EMAIL PROTECTED] ... [default] exten = 2084728803,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) And, finally, the h323.conf: ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay allow=all ; turns on all installed codecs disallow=g723.1; Hm... Proprietary, don't use it... allow=gsm ; Always allow GSM, it's cool :) gatekeeper = 10.0.0.202 AllowGKRouted = yes context=default [2084728803] type=h323 e164=2084728803 context=default Now, including the demo context in default w/out the mgcp extension worked just dandy impressed everyone. I know my box is registering with the gatekeeper just fine the as5300 is getting calles to 2084728803 number to the asterisk box. First: in my [globals] section, is that the right way to use the h323 channel as a trunk? I couldn't find much relating to this out there. Second, what's up with my 8x8 gear? Anyone got it to work? I can attach my mgcp debug output if someone likes. Here's a briefer bit from asterisk -c when picking up the handset on the phone connected to the dta-310: *CLI Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing messa ge from aaln/[EMAIL PROTECTED] tansaction 1 -- Resetting interface aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 2 Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 3 Jan 22 22:07:40 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 4 -- Resetting interface aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 5 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 6 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 7 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 8 -- Resetting interface aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 9 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 10 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 11 Jan 22 22:07:41 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 12 -- Resetting interface aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 13 Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 14 Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 15 Jan 22 22:07:42 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 16 -- Resetting interface aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' Jan 22 22:07:43 WARNING[213006]: chan_mgcp.c:2126 handle_hd_hf: Off hook, but al reaedy have owner on aaln/[EMAIL PROTECTED] Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 17 Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 18 Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 19 Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/[EMAIL PROTECTED] tansaction 20 Jan 22 22:07:43 NOTICE[213006]: chan_mgcp.c:396
RE: [Asterisk-Users] Cisco 7940 with asterisk
This one is for an mgcp phone... In my /tftpboot directory, there's one to match each SEP*.cnf file device devicePool nameDefault/name callManagerGroup members member priority=0 callManager ports analogAccessPort2002/analogAccessPort digitalAccessPort2001/digitalAccessPort ethernetPhonePort3106/ethernetPhonePort mgcpPorts listen2427/listen keepAlive2428/keepAlive /mgcpPorts /ports processNodeName10.0.0.112/processNodeName /callManager /member /members /callManagerGroup /devicePool loadInformationP003E302/loadInformation userLocale nameen/name uid1/uid winCharSetiso-8859-1/winCharSet langCodeen/langCode /userLocale networkLocale/networkLocale idleTimeout0/idleTimeout authenticationURL/authenticationURL directoryURLhttp://10.0.0.112:8086/ciscodirectory?accent=true/directo ryURL idleURL/idleURL informationURL/informationURL messagesURL/messagesURL proxyServerURL/proxyServerURL servicesURL/servicesURL /device -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Gustafson Sent: Tuesday, January 20, 2004 2:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7940 with asterisk Hi all, I've been playing around with a cisco 7940 phone. It seems to like talking to Asterisk with the chan_sccp plugin. The only problem is that it tries to call out to a SEPX.cnf.xml file to verify it's configuration. I've found docs for SEP*.cnf files, but not .xml ones. Does anyone have a .xml file for a 7940 (Skinny?) phone that I can start with? ...Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Residential services
Hi folks, The obligatory newbie disclaimer: Hi, I'm new to Asterisk and I have a couple questions... OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now. Presently, we have: 1) a cluster of sun boxes running propriatary IP-PBX software 2) a cisco 3640 h323 gatekeeper 3) a cisco as5300 pstn gateway I'd like to use sip between an asterisk box and that as5300 (which right now is only speaking h323), and I'd like to be able to use sip, h323, mgcp, or skinny for residential customers. This ought to be no problem, right? I'm coming up with pretty much nil on documentation regarding as5300 - asterisk configuration, however. And, while I'm sure I could fumble through it for a couple days, I thought there just might be someone out there who has a working configuration using an as5300 as a pstn gateway with asterisk (either with sip, or with h323 via a cisco gatekeeper). Now, regarding residential services in particular... The configuration files examples I've found all assume a business environment, where you'd dial a 9 for outside lines. Anyone have an example config where an endpoint gets dumped directly to the pstn when they pick up the phone? Thnaks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Residential services
Thanks all who replied, I think you've gotten me on my way. Over the next few days, while I fiddle with the system I'm testing at home, I'll try to churn out some documentation regarding my setup configuration that may be helpful to someone. I'll submit my notes to the wiki when I'm ready. Anyone else who has residential service and/or cisco interoperability related tips/tricks/real-live-configs (i.e. cisco gateway -- *, cisco gatekeeper -- *), you can reach me off-list if you wish. Thanks Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
