RE: [Asterisk-Users] Wildcard X100P usable in Germany?
Roger. Quick and simple answer. Yes. -Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Roger Schreiter Sent: Monday, January 26, 2004 1:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wildcard X100P usable in Germany? Hi, can I use the X100P in Germany? (I assume I have to exchange some wires.) It would be operated inside a internal PTSN network, so my question is mainly technical, not about permission. Roger. ___ 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] Zapateller
Hi, I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? echo*CLI> show application Zapateller -= Info about application 'Zapateller' =- [Synopsis]: Block telemarketers with SIT [Description]: Zapateller(options): Generates special information tone to block telemarketers from calling you. Returns 0 normally or -1 on hangup. Options is a pipe-delimited list of options. The following options are available: 'answer' causes the line to be answered before playing the tone, 'nocallerid' causes Zapateller to only play the tone if there is no callerid information available. Options should be separated by | characters -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ 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] Using TDM400P for autodial
BTW: Thanks for your help. I had everything configured properly but was working under the assumption that the changes to /etc/asterisk/zapata.conf didn't require an asterisk restart. I guess that explains why I couldn't find the parse logs for zapata.conf when I did a reload. :) Tilghman Lesher wrote: On Sunday 25 January 2004 18:17, Clif Jones wrote: I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have had no luck. The "simple switch" starts every time I pick up the receiver. I have tried this on today's CVS files. Any help and examples would be greatly appreciated. That is indeed how you get it to start in the s extension and not give dialtone. Perhaps you have forgotten that zapata.conf is in object format and the immediate=yes must come before the channel definition? Or perhaps you have forgotten to restart Asterisk after changing zapata.conf? For these situations, it really is helpful for you to post the relevant section of zapata.conf. -Tilghman ___ 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] TE410P on Redhat 9
I am the proud new owner of a TE410P, and installed it on a RedHat 9 box. After compiling just fine, running like a champ in tests, and having my extensions.conf configured to taste, I went ahead and did a live beta test this past weekend. The phone system stopped responding 3 times. The first time, I got into the asterisk console, vi -rv. I didn't see anything too out of the ordinary, except for lots of messages that all of the lines were busy (even though no calls were up and I have 2 PRI's). I thought it odd, but since I had been tweaking a config file or two, I didn't think too much about it. I just stopped * and Zaptel, and then restarted them, in reverse order. And everything was beautiful. THen, on Saturday night, around 7:00 PM, I tried to call in again (I was calling in every half hour to make sure it was working), it had died. I was in a bad mood, and just restarted the same as I had the day before to get it over with. Finally, on Sunday morning, I tested the phone, and at about 8:00 am * was not responding. I logged in, again with 5 v's, and saw an error message to the effect "Span 2 is up". Span 2 is a PRI from * to my Norstar MICS, using a cross-over cable. I think the LBO and signalling are correct, because calls went through it just fine until the 3 times mentioned above. I started by doing a reload, which changed nothing, then a restart when convenient, which changed something... Now, Span 1, 2, 3, and 4 all gave the message that they were up, about 1x per 2 seconds, each. Every 5 or 6 messages I would see that one of the spans was down, but most of the messages were just about being up. I immediately stopped Asterisk and Zaptel, restarted Zaptel, then Asterisk, and everything ran fine through this morning, when I took Asterisk out of the loop (planned end of the beta test). Does anyone have any ideas what might be causing this, and how to diagnose without taking our main phone switch down? I have access to a PRI, but it has no phone numbers that ring into it, so I can connect to real network equipment and make calls, but cannot receive any. Thanks, David Gomillion ___ 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] Re: SIP register/auth with Grandstream BudgeTone-100
/* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ I know it's way off topic, but that is one of the best PS lines I've ever seen! ___ 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] Re: Grandstream 100 sidetone
dkwok wrote: For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? If I'm not mistaken, what you are calling sidetone (the copy of your owm voice that is played back to your earpiece - it's reassuring to hear yourself talk) is actually real echo generated somewhere other than in the phone. It is a network issue, not a phone issue. Read the many, many post on echo and visit the WIKI. ___ 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] Detect Answering Machine in Outgoing calls
I have looked and understand how to create an outgoing call by putting a file in the spool. After searching the archives, I came up with stuff about predictive dialers, but nothing right on topic. I would like to be able to detect an answering machine in outgoing calls so that the message will restart with the beep. In other words, I want to send a message to all of my customers, and if they have an anwering machine, that's OK, I just want to leave a message, the WHOLE message... Has anyone worked on this yet? I know there are a few commercial products that have something they claim will do this, but I think it would be a great feature for Asterisk's outgoing calls. Thanks, David Gomillion ___ 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] Using TDM400P for autodial
Thanks for the advice, sorry that I didn't know what details you needed. ;) The format does come before the channel and I even tried moving it. However, I didn't restart Asterisk because I figured that since everything else appeared to reconfigure on a "reload" command that I didn't have to. Nice little surprise. I will try restarting asterisk. Wonder what else under /etc/asterisk doesn't get reloaded upon the "reload" command??? Tilghman Lesher wrote: On Sunday 25 January 2004 18:17, Clif Jones wrote: I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have had no luck. The "simple switch" starts every time I pick up the receiver. I have tried this on today's CVS files. Any help and examples would be greatly appreciated. That is indeed how you get it to start in the s extension and not give dialtone. Perhaps you have forgotten that zapata.conf is in object format and the immediate=yes must come before the channel definition? Or perhaps you have forgotten to restart Asterisk after changing zapata.conf? For these situations, it really is helpful for you to post the relevant section of zapata.conf. -Tilghman ___ 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] app_queue and dialplan
Try with: http://bugs.digium.com/bug_view_page.php?bug_id=214 Regards, Gus - Original Message - From: "Anton Yurchenko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 26, 2004 9:59 AM Subject: [Asterisk-Users] app_queue and dialplan > Hello, > > I`m trying to achive this: > 1. when the initial call comes in it is served by a small queue with > short timeout so that at first caller hears only ringing > 2. if nobody answers the call at that time or the queue is all full the > call goes to the Playback the message ( "please hold bla bla bla") > 3. Then the call goes to another queue and he holds while the > music-on-hold plays a app_queue trys to reach the next free operator > 4. after a timeout in second queue there is a Goto to play the message > again and then back into the second queue > > > I have it like this: > > extensions.conf: > exten => 10,1,Queue(q1_short,tn) > exten => 10,2,Answer > exten => 10,3,Playback(please_hold) > exten => 10,4,Queue(q1,t) > exten => 10,5,Goto(3) > > > queue.conf: > > [q1] > music = test > announce = test_anounce > timeout = 40 > retry = 3 > maxlen = 10 > > strategy = leastrecent > > member => SIP/111 > member => SIP/112 > member => SIP/113 > member => SIP/114 > member => SIP/115 > > > [q1_short] > music = test > announce = test_anounce > timeout = 15 > retry = 3 > maxlen = 3 > strategy = leastrecent > member => SIP/111 > member => SIP/112 > member => SIP/113 > member => SIP/114 > member => SIP/115 > > > but the broblem is when the q1_short is full, and the call goes to the > q1 it only plays the announce message and and no music on hold is played > and again the announce message is played. somehow the music on lod > doesn start. What am I doing wrong? > I run version CVS-12/01/03-14:50:57 > > Thanks > > > -- > > Anton Yurchenko<[EMAIL PROTECTED]> > Digital Generation > > > ___ > 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] GSM modems
Quoting Steve Underwood <[EMAIL PROTECTED]>: > I am interested in interfacing a GSM modem to *. I've seen a few > comments about doing this, but I'm not clear whether people have > actually made it work. You could try to search for ATEUS by www.2n.cz or Ecotel There were also some cheap (about 200US$ for BRI) from taiwan but i cant find the url. rgrds m. ___ 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] My experience with one IAX termination provider and one SIP provider
In the wake of the recent NuFone thread I'd like to offer up a brief summary of my limited experience with both Vonage and VoicePulse Connect. I live in Houston TX and for about a year had a Vonage line as my entre into VOIP. Back then, as now, I'd go to some length to avoid Southwestern Bell. I work for a UK based company and my operational territory is "The Americas" thus I use a lot of long distance, mostly to the US, Canada & the UK. The Vonage service was very reliable and I could usually not tell the difference between one of my POTS line and the ATA connected Vonage line. Their service was in my opinion exemplary. One problem with Vonage for me is the billing model. When I travel (a lot) paying a monthly fixed price makes no sense. Some months I use very few minutes, others 2000+. Also, I couldn't get a real second line without incurring another significant monthly fee. Still, they offered local DIDs and I used them happily for a year. In looking at * I was attracted by the prospect of control, features, moh, aa, etc. So I built an * server and bought a few SNOM phones. This displaced my 4 line/8 ext Panasonic office phone system. Connecting the Vonage provided ATA to an X100p didn't seem to make much sense. Needless D/A and A/D steps. Passing SIP through my router meant opening up multiple ports which I really didn't want to do. So I sought out a provider who offered IAX terminationand ended up with VoicePulse Connect (VPC). I've only been using VPC for about a month. The quality of calls over VPC is not quite as good as I had with Vonage. I suppose that could be in part a codec issue. At present I use GSM to connect to VPC to save some bandwidth. I'm open to suggestions as to how to tweak the connection. The service was easily setup. From the point of account creation to initial call was about 30 minutes, largely due to my having to edit the dialplan. No waiting for hardware to ship. The account was up and ruinning immediately even as I worked on it near midnight. The service has thus far been reliable. It supports multiple (up to 6) simultaneous outbound calls which I have used to conference several co-workers. The management imterface on their web site is not as feature rich as Vonage, but it gets it done for me. I like the pay as go, no monthly fee billing model. It's so convenient that even if I use another provider I will keep VPC as a fallback/alternate in my dialplan. One problem however is their lack if DIDs in my area, so I only use it for outbound calling. POTS lines for inbound. Later this month I will likely drop two of my incomming POTS lines as the * server takes on more of the load. That's a vote of confidence in VPC, but I'm still looking for options. If NuFone provides Houston DIDs then they're likely my next experiment. Broadvox Direct, while interesting, is a non-starter due to their flat rate billing model ala Vonage. I also hope to add a wifi sip phone to fill in the last void (cordless) left by the departure of the Panasonic system...if ever they become available. I am eager to hear about others experience with the various providers, even IP Centrex providers. This is just my recent experience. As usual YMMV. Michael Graves P.S. - what to do with my Vonage-crippled ATA? -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] "With us or against us isn't a policy worthy of a democratic superpower." -- Zbigniew Brzezinski, Former US National Security Advisor ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Scalability/Benchmarks/Performance
Any Scalability/Benchmarks/Performance data? Digging through the archives I found some questions but no answers. Does anyone have a BHCA (Busy Hour Call Attempt) value for an asterisk setup (please also include your hardware config). Even a raw call generation number would be greatly appreciated. Thanks Javed
Re: [Asterisk-Users] rc.local dont works
On Mon, 26 Jan 2004, Jeroen wrote: > Hi Miklos, > > I have the same problem here in RH90 - have you found any solution? > > Or does anybody else know why (safe_)asterisk does not start using > rc.local? (normally I start * as root user) > > Cheers > Jeroen Jeroren, I've included an asterisk.init in my RedHat 9.0 RPMS. You can grab them from ftp://ftp.nacs.net/asterisk/ -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [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 behind NAT - use of "externip" option
I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[EMAIL PROTECTED]/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP Debugging Enabled Jan 26 18:20:04 NOTICE[9226]: chan_sip.c:3126 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again 11 headers, 0 lines Reliably Transmitting: REGISTER sip:voiptalk.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.89:5060;branch=z9hG4bK02c0373f From: ;tag=as5548d275 To: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: Event: registration Content-length: 0 (no NAT) to 82.145.32.73:5060 Retransmitting #1 (no NAT): REGISTER sip:voiptalk.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.89:5060;branch=z9hG4bK02c0373f From: ;tag=as5548d275 To: Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: Event: registration Content-length: 0 to 82.145.32.73:5060 Retransmitting #2 (no NAT): ... as previous frame ... I would expect (perhaps wrongly?) that the externip would be reflected in the Via: header (it seems to be when I look at the traces from a SNOM 100 successfully registery with voiptalk behind the same firewall). And the (no NAT) comments in the debug trace also look suspicious, given the explicit nat=yes in the config. Any hints? I guess the externip might be in the wrong place or conflicting with one of my other options, but neither the docs don't seem to offer much by way of advice (I've checked the wiki, googled etc). Thanks Patrick ___ 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] OH323 doesnt hear ringing
I am using H323 not OH323
and have the same problem.
My h323.conf :
; The NuFone Network's
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = xx.xx.xx.xx
;tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel calls. It must be
; one of 'default', 'omit', 'billing', or 'documentation'. These flags
; are used in the generation of call detail records.
;
;amaflags = default
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
;accountcode=lss0101
;
; You can fine tune codecs here using "allow" and "disallow" clauses
; with specific codecs. Use "all" to represent all formats.
;
;allow=all; turns on all installed codecs
;allow=g723.1 ; Hm... Proprietary, don't use it...
;allow=g729 ; Hm... Proprietary, don't use it...
allow=gsm ; Always allow GSM, it's cool :)
allow=ulaw
allow=alaw
;
; User-Input Mode (DTMF)
;
; valid entries are: rfc2833, inband
; default is rfc2833
dtmfmode=inband
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
; or - The acutal IP address or hostname of your GK
gatekeeper = DISABLE
;
;
; Tell Asterisk whether or not to accept Gatekeeper
; routed calls or not. Normally this should always
; be set to yes, unless you want to have finer control
; over which users are allowed access to Asterisk.
; Default: YES
;
AllowGKRouted = yes
;
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
context=blah
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls [EMAIL PROTECTED]
; Asterisk will send the call to the extension 'time'
; in the context default
;
;[default]
;exten => time,1,Answer
;exten => time,2,Playback,current-time
;
; Keyword's 'prefix' and 'e164' are only make sense when
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias 'time'.
;
;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
[blah]
type=friend
host=xx.xx.xx.xx
context=default
exten=>,1,Goto(s,1)
incominglimit=4
prefix=9
busydetect=yes
allow=ulaw
allow=alaw
- Original Message -
From: "Michael Manousos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 26, 2004 10:27 AM
Subject: Re: [Asterisk-Users] OH323 doesnt hear ringing
> Aaron Martin wrote:
> > I have Asterisk running with a combination of SIP and H323 clients. I
> > am using the OH323 module instead of the H323 one.
> >
> > When the SIP clients ring each other, they can hear a ringing noise in
> > the ear peice to let them know that the other parties phone is ringing.
> > However, when the H323 client rings a SIP client, there is no ringing
> > sound at all, although as soon as the called party picks up the phone,
> > everything works fine. This is the entry from my extensions.conf:
> >
> > exten => _7[5-9]X,1,Dial(SIP/${EXTEN},20,rt)
> > exten => _7[5-9]X,2,Playback(vm-nobodyavail)
> > exten => _7[5-9]X,3,Hangup
> >
> > I assume that because I havr the 'r' in the dial string, the calling
> > party should hear a ringing noice. Any ideas?
>
> What are the contents of your oh323.conf? (please hide
> passwords/IPs, if any).
>
> Michael.
>
>
> ___
> Asterisk-Users mailing list
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>
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Re: [Asterisk-Users] Questions regarding new echo cancellation features...
On Monday 26 January 2004 07:53, john wrote: > I notice the zaptel Makefile option > the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR > is now gone. Does simply adding these options still compile in a > certain echo can - or is there an other method of activating a > particular can. I have not had to update my machine that is > connected to pstn for a while & I don't want to jump into > echotraining without a way to quickly enable what has been working > for me. > How do I enable this mode now? If you examine the Makefile a little, you'll find that all such options have been moved to zconfig.h. -Tilghman ___ 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: Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
> I'll be trying that as my next step but it seems that my other fresh - > sipura 2000 unit that was sitting in the box which is running 1.0.15 > firmware seems to work seamless so i find it odd ? that brand new unit > works while the upgraded firmware ones don't? > > I'm not the only one having this same exact issue I've received 4 > emails relating to the same issue from other users. So there's some > kind trend going on with this? Unless someone else just happens to have gone through the exact same thing (probably not all that likely), it won't help to keep posting general statements. Really need to see "something" else like some output from debug, packet flows, etc. Since I don't have a sipura, I would not have a clue whether "sip debug" would provide anything, but I'd bet a few on this list might be able to interpret an etherial packet trace. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk RPMS Updated (FC1,RH9,RH73)
Hello all, On Sunday morning MER-B (known as Opportunity) successfully entereed the Martian atmosphere, descending through "5 minutes of hell" and eventually bouncing to a stop at Meridiani Planum a little after midnight. I stayed watching NASA TV and to celebrate, I have released updated Asterisk 0.7.1 RPMS. The RPMS are now at release 3, and I have added support for Fedora Core 1 thanks to feedback from William Lorenz. However, I need more testers and feedback, and YOU are the perfect person to tell me what you would like to see! Please download, test, break and suggest updates and fixes! I have updated and will be maintaining a page on the Asterisk Wiki located at: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM Current Release --- asterisk-0.7.1-3.i386.rpm libpri-0.5.1-3.i386.rpm zaptel-0.8.0-3.i386.rpm kernel-module-zaptel-0.8.0-3_2.4.20_28.7.i386.rpm FTP Download ftp://ftp.nacs.net/asterisk/ Changelog - * Mon Jan 26 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - Updated changelog entry to enable build on Fedora Core 1 <[EMAIL PROTECTED]> - Made the decsision to use Dist Specific version numbers (_fc1,_rh9,_rh8,_rh73) * Sat Jan 24 2004 Gregory Boehnlein <[EMAIL PROTECTED]> - added doc macros - added config macros - updated install stanza to correct symlink issue - updated patch0 to include changes to Makefile - added /etc/rc.d/init.d/asterisk - added "export LD_ASSUME_KERNEL=2.4.1" for RH9 - asterisk.spec now builds cleanly on RH73 and RH9 * Wed Jan 21 2004 Gregory J. Boehnlein <[EMAIL PROTECTED]> - Initial .spec file created. Most likely buggered. Badly needs help. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [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] ZAP Problems
I would say it might be this... n zapata.conf language=en contect=default -> should be context=default switchtype-euroisdn signaling=fxs_ks rxwink=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Posted At: Monday, January 26, 2004 6:12 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] ZAP Problems Subject: [Asterisk-Users] ZAP Problems Hi all, Since my upgrade to CVS dated 14-01-2004 I am unable to call or receive calls through my ZAP channel. When calling out I get the following message: - WARNING [155667]:app_dial.c:527 dial_exec: Unable to create channel of type ZAP In zaptel.conf fxsks=1 loadzone=uk defaultzone=uk In zapata.conf language=en contect=default switchtype-euroisdn signaling=fxs_ks rxwink=300 I have done: - modprobe zaptel modprobe wcfxo ztcfg -vv results: - Zaptel Configuration Channel Map: Channel 01: FXS Kewlstart (Default) (Salves:01) 1 Channels configured Any help to resolve would be appreciated. Regards Dave ___ 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] 7960 Problems
This is not specifically related to * but * is the software I’m using so here goes… Does anyone have the correct file set for a 7960?? I’ve been trying to get the release 6 SIP load on one I have without any luck. The phone keeps getting the same 2 files from the tftp server and starting over. If you have the files – other than the POS30600.bin which I know is licensed – could you please send them to me so I can figure out if it’s my files or my phone?? I really would appreciate any possible help with this. Thanks, Lane Hoskins, MCP Network Engineer 540.767.7626
