Re: [asterisk-users] help with DAHDI hangup on calling out.

Jerry Geis wrote: > I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri > 1.4.7 > > and I am getting the error: > -- Requested transfer capability: 0x00 - SPEECH > -- Called 23/317506 > -- Channel 0/23, span 1 got hangup, cause 99 > -- Hungup 'DAHDI/23-1

Re: [asterisk-users] Asterisk quits responding

Asterisk still does things like use DNS SRV records unless turned off, it also tries to do a DNS lookup for all IP addresses on the system. Update your hosts file and see if that helps. Todd Reese wrote: > DNS server rebooted. Everything is back online now. > > What threw me was that one of t

Re: [asterisk-users] Asterisk quits responding

Chances are you have a DNS problem. Asterisk is trying to look up a hostname and it is not getting a response. Todd Reese wrote: > > New update: After the console being frozen for about 15 minutes, it > just responded to the last reload comand >

Re: [asterisk-users] 1.6.1-rc4: extension "i" not working??

sean darcy wrote: > I've have a simple caller id lookup on incoming: > > [teliax-in] > .. > exten =>s,n,GoSub(set-callerid-name,0${CALLERID(num)},1) > > > [set-callerid-name] > exten => 0,1,NoOp( no CALLERID num set) > exten => 02135590993,1,Set(CALLERID(name)=Matthew )

Re: [asterisk-users] SMS text messaging capabilities

Elliot Murdock wrote: > Hello! > > What kind of sms text messaging capabilities does Asterisk have? Asterisk has the ability to use land lines to send SMS messages to a remote device that supports landline SMS. In Europe and much of the rest of the world SMS carriers provide a public PSTN ga

Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem

obitori junk wrote: > I am experiencing a "606 not Acceptable" error trying to set up an > Asterisk server as an ekiga.net client. My server is behind a firewall > with NAT routing. I have googled this problem and read about Asterisk > feeding its local ip address to ekiga.net. That seems to be

Re: [asterisk-users] Cut Through DTMF & caller ID on SIP phone

Sriram wrote: > Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed > that whenever during "background(menu-filename)" method - i try to > press any key for selection like 1 for some prompt, 2 for another > prompt etc...Asterisk takes a while before it takes me to the > respective o

Re: [asterisk-users] Increase DTMF Tone Duration

Andres wrote: > We are running 1.4.22 and have been experiencing problems with certain > IVRs and DTMF Tone duration. We would like to be able to increase DTMF > Tone duration by 50 to 100ms over what the user is pressing on his > phone. We have a PRI test circuit and an analyer in between to

Re: [asterisk-users] top posting again [was: Re: CDR Design]

Barry L. Kline wrote: > Bill Andersen wrote: > >> In the "order in which people normally read text" they don't >> repeat the entire conversation from the beginning each time >> a question is asked either... Bottom posting is just as bad! >> >> ./bill > > Not when you take the time to properly t

Re: [asterisk-users] Dedicated Fax Line

In the USA (maybe other T-1 countries) you can have a channelized T-1. Each channel is assigned signaling just like an analog line, FXS, FXO, E&M, etc. I have worked with a carrier in the past that could put FXO channels on a T-1 along with a PRI channels on the same T-1. Andrew Thomas wrote:

Re: [asterisk-users] Follow up on parking

body) because of the dynamic park-dial context. > > Regards, > > Mike > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric > "ManxPower" Wieling > Sent: Monday, Dec

Re: [asterisk-users] Follow up on parking

Terry Wilson wrote: > On Dec 15, 2008, at 7:05 AM, Mike wrote: > >> Just so I'm clear: there is no way to do what I want short of >> playing with the underlying code, correct? > > Yes. I'm working on an issue right now related to parking and noticed > that Asterisk completely lies with the

Re: [asterisk-users] ring back tone

Philipp Kempgen wrote: > michel freiha schrieb: > >> I would like to ask please if there is a way to play a ring back tone from >> asterisk when the customer try to make a call...I already added the ringing >> function to the context in extensions .conf and it work perfectly...But the >> issue t

Re: [asterisk-users] ring back tone

No, not on FXO ports. On FXO ports Asterisk considers the call answered as soon as dialing is finished. Asterisk has no way to detect when the far end answers when using FXO ports. michel freiha wrote: > I would like to ask please if there is a way to play a ring back tone from > asterisk when

Re: [asterisk-users] call to mobiles and it is turn off

Remove the "r" option to Dial. Bruno Castelo Branco wrote: > Hi all > > When I call to any mobile and the device is power off the asterisk keep > ringing and I not able to hear the tradicional message saying this > mobile is power off. > When I call from a normal analogic line I got the

Re: [asterisk-users] Parked Extension Variable

Use the docs, Luke. dev-1*CLI> core show application parkandannounce dev-1*CLI> -= Info about application 'ParkAndAnnounce' =- [Synopsis] Park and Announce [Description] ParkAndAnnounce(announce:template,timeout,dial[,return_context]): Park a call into the parkinglot and announce the call

Re: [asterisk-users] Asterisk variable for SIP context

Mike wrote: > Sure, that works too, but I needed access to context taken from the sip > entry because I needed to goto(${that_context}), and that context varies > depending on the phone used. > > This is when setting __TRANSFER_CONTEXT in my cookie cutter really > big/complexe/database driven ex

Re: [asterisk-users] Packet size limit for HDLC?

ICMP is used to determine maximim packet size. If you or the other end are blocking all ICMP then MTU Path Discovery will not work. It's a classic newbie network admin mistake. Symptoms of this problem would be exactly like you describe. Typically I see this on PPPoE connections. More info:

Re: [asterisk-users] Call parking

Welcome to the world of FreePBX. It would save me quite a bit of time if you could list what ports (port number and signaling) you have on the card and what context you want each port to go into. When I manually merge the two files (after stripping out 37 billion comment lines) I see that cha

Re: [asterisk-users] We think we are cpe but they think they are cpe too

Next time I'll be sure to finish my morning coffee before posting. 8-) Tony Mountifield wrote: > In article <[EMAIL PROTECTED]>, > Eric \"ManxPower\" Wieling <[EMAIL PROTECTED]> wrote: >> There is a loopback somewhere on the line. Contact your telco an

Re: [asterisk-users] canreinvite=yes problem

gt; > I need that sip sign go to Asterisk2 > But RTP go to Asterisk1 and no more. > > Where have I to insert canreinvite ? > > Thank you > > > > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part de Eric > &qu

Re: [asterisk-users] We think we are cpe but they think they are cpe too

There is a loopback somewhere on the line. Contact your telco and say "I see a loopback on the line. Please remove it." Uros Djokic wrote: > Hi I have problem with TE121 Digium card. I connected it to modem keymile > Music 200 (provided by telco) but I can see 2 red lights on modem (both > bell

Re: [asterisk-users] Call parking

- isn't this an attended transfer? > > on Wednesday 12/03/2008 "Eric \"ManxPower\" Wieling"([EMAIL PROTECTED]) wrote > > > > John covici wrote: > > > OK, I can park the calls OK, but I don't get the announcement -- I am > > >

Re: [asterisk-users] Call parking

John covici wrote: > OK, I can park the calls OK, but I don't get the announcement -- I am > using freepbx if that makes any difference. If you park a call and do not hear the announcement then you are doing a BLIND transfer, not an ATTENDED transfer. You should be doing attended transfers for

Re: [asterisk-users] canreinvite=yes problem

canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by

Re: [asterisk-users] Parking calls

It is not a parking solution. Sebastian wrote: > Any idea? Please I need advice. > > Thanks! > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian > Sent: lunes, 01 de diciembre de 2008 11:58 p.m. > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > S

Re: [asterisk-users] pick up IAX2 calls

You can either add that feature to chan_iax2.c or pay someone to add that feature to chan_iax2.c. Bruno Castelo Branco wrote: > Somebody know some work around for it? > I still trying to find a solution but nothing seems to work > > thanks > > Eric "ManxPower" Wiel

Re: [asterisk-users] pick up IAX2 calls

The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: > hi > I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 > for all IAX extensions in iax.conf. Didn't works for while. > thanks > > Tim Panton wrote: >> I think it doesn't work across

Re: [asterisk-users] pick up IAX2 calls

Bruno Castelo Branco wrote: > Somebody knows if pickup call works with IAX2? > I enable *8 in features.conf, but doesn't works with IAX2 extensions. As I understand it, IAX2 does not support callgroup= and pickupgroup= and *8. This link might be helpful: http://www.freepbx.org/trac/ticket/15

Re: [asterisk-users] sip trunking and call transfer

Maybe because there is no such thing as a "SIP trunk", at least in the Asterisk world. Most of us call them "peer" or "friend". The term you are looking for is "reinvite". Reinvites allow two devices to send audio directly between the two end points of the call. the SIGNALING stays on the se

Re: [asterisk-users] Using MAC or extension number as SIP identifier

Olivier wrote: > 2008/11/20 Eric ManxPower Wieling <[EMAIL PROTECTED]> > >> Personally I use the MAC-x wherex=the line appearance number. MAC-a for >> first line appearance, MAC-b for 2nd, etc. >> > > Is it easy to use (CLI, logs...) ? > Wo

Re: [asterisk-users] Using MAC or extension number as SIP identifier

Philipp Kempgen wrote: > Olivier schrieb: > >> For a long time, I was wondering if I should use MAC address instead of >> Extension number to identify SIP endpoints (as I'm mostly not using >> softphones). >> >> Before diving into this, I wondered how people using MAC address are using >> CLI as

Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

Tilghman Lesher wrote: > On Wednesday 12 November 2008 23:15:01 Eric "ManxPower" Wieling wrote: >> Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and >> chan_iax (maybe other channels) translate the protocol specific causes >> to a Q.931 hangup

Re: [asterisk-users] How to get correct dial result for outgoing calls thru ISDN?

Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and chan_iax (maybe other channels) translate the protocol specific causes to a Q.931 hangup cause. I would recommend you check the doc directory in the Asterisk source code for channelvariables.txt, but for some reason that I ca

Re: [asterisk-users] Why Nat=yes Nat=no Option?

Alex Balashov wrote: > Steve Totaro wrote: > >> I have done some large installs where people are going to be in the >> office, sometimes out, work from home, it always changes sorta thing.. >> >> I have found that setting all device profiles to Nat=yes "Just Works" >> whether they are on t

Re: [asterisk-users] E1 PRI to and from SIP screeching

From IRC: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc),

Re: [asterisk-users] Codec problems when using G.723

Thomas Winter wrote: > On Sunday 09 November 2008 20:14, Eric "ManxPower" Wieling wrote: >> The best (and maybe only way) is to set your client and your service >> provider to only do G.723. > > Really, thats not the way it should work. > > How I can fi

Re: [asterisk-users] Recommend Wireless IP Phone

You can sometimes find the older Cisco Aironet boxes that run at 900Mhz. That frequency is AWESOME in rural areas. Mountains will still block it, but trees and water does not. Drew Gibson wrote: > Wilton Helm wrote: >> Good points. I got an access point instead of a router specifically >> s

Re: [asterisk-users] Codec problems when using G.723

The best (and maybe only way) is to set your client and your service provider to only do G.723. Thomas Winter wrote: > Hi, > > I have a problem with codecs. > > I have an provider with allowed codec alaw, ulaw, g.723 > I have SIP clients with codec allowed alaw, ulaw, g.723 > > If a SIP client

Re: [asterisk-users] twice normal beep before busy tone ??

Most IVRs want longer DTMF tone lengths. If you shorten the toneduration= then many IVRs won't work. Wilton Helm wrote: > If it is 300 ms, that is way to long. I don't know any CO grade receiver > that can't decode in 80 ms and some can do 40. There is also a similar size > gap between digit

Re: [asterisk-users] twice normal beep before busy tone ??

It sounds like you have analog lines. If that is the case, the silence you experience is Asterisk sending the DTMF down the line. Asterisk collects the DTMF and when you are done dialing it retransmits those digits down the analog line. I think each digit is by default 300ms. If you are dial

Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf

Historically Asterisk's config file parser ignored unknown keywords. This is useful for exactly the things you are trying to do. I hope 1.6 did not remove this "feature". Rob Hillis wrote: > Barry L. Kline wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Rob Hillis wrote: >>

Re: [asterisk-users] Asterisk installation

Then you should read the READMEs right now. See the 3 upgrade info files as well as any other READMEs. Christian wrote: > Hello, > Many thanks for the info. > OK, I didn't know that. I just installed it. Usually I read the included read > me files and so on but not at this time. > But I will be

Re: [asterisk-users] SIP # DTMF

"core show application dial" (this is the official application doc) Pay special attention to the D() option. Rodolfo Alcazar Portillo wrote: > Hi. In creating a custom extension, and dialing > > SIP/222/333#444, seems the party receives only "333" > > What should I do to send the # symbol? o

Re: [asterisk-users] Dealing with progress codes

From zapata.conf.sample: ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONN

Re: [asterisk-users] adding a second extension

ast_request: No channel type registered for ''SIP' Notice the extra ' in the message. That is either an error in the error message or you have a an extra ' in your Dial line. Something like Dial('SIP/ I'm surprised nobody else noticed this. Stephen Reese wrote: > On Sun, Oct 19, 2008

Re: [asterisk-users] srv records not being honoured properly

Brian J. Murrell wrote: > On Fri, 2008-10-17 at 11:18 -0500, Eric "ManxPower" Wieling wrote: >> It should be fairly easy to write an AGI script that does the SRV query, >> do whatever you want with the response, set a channel variable with >> the result

Re: [asterisk-users] srv records not being honoured properly

It should be fairly easy to write an AGI script that does the SRV query, do whatever you want with the response, set a channel variable with the results and use that in your dialplan. Brian J. Murrell wrote: > On Fri, 2008-10-17 at 10:32 -0500, Andres wrote: >> Because Asterisk does not suppor

Re: [asterisk-users] Strip prefix

exten => _+X.,1,Goto(${EXTEN:1},1) michel freiha wrote: > Dear All, > > i have the following context defines in etensions.conf: > > > [a2billing] > exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) > exten => _X.,2,DeadAGI,a2billing.php > exten => _X.,3,Wait,2 > exten => _X.,4,Hangup > exten => _X

Re: [asterisk-users] Matching *, + and # in the dialplan

exten => +13129842314,1,Noop(Happy match!) or exten => _+1NXXNXX,1,Noop(Happier match!) Karl Fife wrote: > "Steve Murphy" <[EMAIL PROTECTED]> wrote: >> People have voiced this before; but the cut-down version of RE's that >> the matching algorithms allow are fairly fast, both in the new an

Re: [asterisk-users] Question about echo cancelation

normal echo canceling systems. Most echo canceling systems I've seen (mostly tellabs) only cancel echo in one direction. I suspect all of Digium's EC systems only do echo canceling in one direction as well. Olivier wrote: > 2008/10/11 Eric ManxPower Wieling <[EMAIL PROTECTED]&g

Re: [asterisk-users] 1 second delay when connecting calls

Try setting canreinvite=no in each of the device sections on a couple of phones, reload chan_sip.so and see if that fixes things. It has fixed the issue when I've tried it. [EMAIL PROTECTED] wrote: > Hello, > > We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we > make

Re: [asterisk-users] Question about echo cancelation

Olivier wrote: > 2008/10/10 Eric ManxPower Wieling <[EMAIL PROTECTED]> > >> All calls with a 2-wire analog piece have echo. You cannot perceive the >> echo because it happens so fast on non-VoIP connections. On VoIP calls >> you have significant extra latency wh

Re: [asterisk-users] Question about echo cancelation

All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Echo must be removed before the call is converted to VoIP -- in your ca

Re: [asterisk-users] Got event 17 (Polarity Reversal)...

You should not get that message on analog lines in the USA or Canada. I suspect your line has a provisioning issue or is using different signaling than you think it is using. Jim Duda wrote: > Can anyone tell me what this message means? > > Got event 17 (Polarity Reversal)... > > I'm running

Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]

) write a script that polls > subversion docs directory and imports it into voip-info.org when the the > docs are changed? > > I'd be glad to write and host such a script if the community desires the > feature. > > -josiah > > SIP wrote: >> Eric "ManxPo

Re: [asterisk-users] ATA for large networks

The most reliable "ATA" is a channel bank. Vieri wrote: > --- On Mon, 9/29/08, Sam Tam <[EMAIL PROTECTED]> wrote: > >> Why not swap it all with just IP phone? > > That's because we have almost 400 analog phones already wired in our building > (which is very large). So we need to take advantage

Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]

Olivier wrote: > I don't have any spare zaptel enabled system I could try this on, but I > was not aware of this CHANNEL variable. > Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables > Maybe, I will add a line in www.voip-info.org > to keep

Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX

I believe chan_iax2 does not support call pickup. Search the archives. Shazaum wrote: > already tested with an exten? > ex: > exten => _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED]) > exten => _*8.,n,Hangup() > > 2008/9/27 coco <[EMAIL PROTECTED] > > > Hello list > >

Re: [asterisk-users] extension definition

This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the time a call gets to extensions.conf it must already be authenticated. Assume the username is robertdobbs and the ip is 209.17.71.61 In sip.conf you would have something like this: [robertdobbs] deny=0.0.0.0/0 permit=209.17

Re: [asterisk-users] Digium training course

Tilghman Lesher wrote: > On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: >> On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno <[EMAIL PROTECTED]> wrote: >>> Anybody knows how to get a Coupon Code for the discount on the Asterisk >>> training classes??? I am interested on taking that upco

Re: [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle

Where did you hear this? Shaun Wingrin wrote: > I have heard it said that, Asterisk falls over at 100 simultaneous > calls. Is this true? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL.

Re: [asterisk-users] SCCP port numbers used for audio stram?

SCCP (aka Skinny), H323, MGCP, and SIP all use the RTP protocol for audio. For all signalling protocols (except maybe H323) use rtp.conf for the RTP ports. OCG Technical Support wrote: > > > I have a 7921 wireless phone working with Asterisk, and I want to > tighten the wide open port range

Re: [asterisk-users] SIP Extension Config Issue

You're joking, right? exten => _1xx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr) exten => _1xx,n,Voicemail([EMAIL PROTECTED]) Use whatever voice mailbox and voicemail context you want. Joseph L. Casale wrote: > I have a setup with a SIP DID inbound, and several SIP phones insid

Re: [asterisk-users] FAX over T1 Question

practical for a PRI then? >> >> Thank you!!! >> >> ++Amaru >> >> --- On Fri, 9/5/08, Eric "ManxPower" Wieling <[EMAIL PROTECTED]> wrote: >> >> From: Eric "ManxPower" Wieling <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-

Re: [asterisk-users] FAX over T1 Question

The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me implies Channelized T-1). Seems like most people gave you advice that might apply to a Channelized T-1, but would not apply or be practical for a PRI. Amaru Netapshaak wrote: > Bob, > > I should have added that I have disa

Re: [asterisk-users] FAX over T1 Question

If I am not mistaken every single echo canceler out there will disable itself if it detects a fax tone. Echo Cancelers do not screw up faxes, people screw up faxes. 8-) Bob Pierce wrote: > On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote: >> I have a Sangoma A104d T1 card, a Rhino 24-po

Re: [asterisk-users] ringback when the channel is answered

ption it it right? > > > > 2008/9/4 Eric ManxPower Wieling <[EMAIL PROTECTED]>: >> It will do so by default if you have a valid >> /etc/asterisk/indications.conf (only used for inband tones like after an >> Answer()) >> >> eng. Anatoli Marino

Re: [asterisk-users] ringback when the channel is answered

It will do so by default if you have a valid /etc/asterisk/indications.conf (only used for inband tones like after an Answer()) eng. Anatoli Marinov wrote: > Hi guys, > I am trying to configure an asterisk server for our office. > Asterisk 1.4.17 SIP only > > The problem appears when the call c

Re: [asterisk-users] X100P Card in OFFHOOK state

It would be clearer if it said "Hookstate (FXS ports only): Offhook" i.e. the state information is not valid for FXO ports. Jay Ray wrote: > Any pointers on this one? > > --- On Tue, 8/26/08, Jay Ray <[EMAIL PROTECTED]> wrote: > From: Jay Ray <[EMAIL PROTECTED]> > Subject: [asterisk-users] X100P

Re: [asterisk-users] set callerid with plus sign

+ is not a valid Caller*ID character. Asterisk allows you to use + in Caller*ID, but many carriers will reject the call if you do that. Benny Amorsen wrote: > ronald <[EMAIL PROTECTED]> writes: > >> Is it possible to assign a plus sign on the callerid(num) ? > > Yes. > >> currently this is wh

Re: [asterisk-users] 5 min limitation on phone calls! how to!

Steve Totaro wrote: > On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd <[EMAIL PROTECTED]> wrote: >> Hello all! >> >> my last month's phone bill sky rocketed after i setup asterisk with >> softphones all over the house! >> >> could someone help me set up a limitation for my wife and kids not to be

Re: [asterisk-users] Perl AGI defunct process

Your script is not catching SIGHUP, which is what Asterisk uses to tell the AGI the channel went away. Ruddy Gbaguidi wrote: > Hi all. > I'm using asterisk 1.4.21.2 and when I run > ps -ef |grep defunct, > I can see a lot of my perl agi still pending there. > The channel has been cleaned up in as

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

But what would you call it? It's not a card, so it can't be a "NIC", right? Steve Totaro wrote: > Er, it would be one integrated with the MoBo, on the board if you will... > > Thanks, > Steve Totaro > > On Thu, Aug 14, 2008 at 11:50 AM, Eric ManxPow

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

What would "on-board NIC" be? Jay R. Ashworth wrote: > On Wed, Aug 13, 2008 at 11:54:23PM -0400, Steve Totaro wrote: >> "NIC card" is redundant ;-) > > And you can take that to the ATM machine. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, L

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

Steve Totaro wrote: > On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller <[EMAIL PROTECTED]> wrote: >> From what I can determine while troubleshooting a voice-dropping >> issue, the Asterisk server in my organization has been dropping RTP >> packets between the asterisk server process and the netwo

Re: [asterisk-users] re-distributing E1

Yes you should be able to do that with an E-1 (it's called "DACS"). HOWEVER, you can't do DACS on a PRI, as you would need the D-Channel replicated and you can't do that. If you just want Asterisk to provide PRIs to your users, then that's a different story. Hans Witvliet wrote: > Before tryin

Re: [asterisk-users] Acceptance testing of a new PRI

The "something" is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means "first instance of a call on this channel", a -2 would be see

Re: [asterisk-users] IAX to work on two ports: 4569 and 4570

bilal ghayyad wrote: > The reason that I need to do this is: > > I will have two Asterisk PBX's, and I need both of them to use same Internet > (so both of them will be behind NAT under same DSL router), in that case, how > I will distinguish on the router the calls that need to be send for bo

Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6

I sit corrected. He should still be reading the upgrade files. Doug Lytle wrote: > Eric "ManxPower" Wieling wrote: >> >> It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that >> should have been included in the source code. If you re

Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6

Jerry Geis wrote: > I dont see any errors in the dialplan while loading. > I tried to past the whole log but it was rejected. > > Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files. > > I cant even "dialplan show default" at this time. It looks like you did not read the UPGRA

Re: [asterisk-users] GotoIf Problem

How about: exten => _9X,n,Goto(not-parked,s,1) Doug Lytle wrote: > Everybody, > > I have a fall though context that, among other things, tests to see if > someone it trying to pick up a non-existent parked call (Defined from 90 > to 99). I have the following: > > [not-in-service] > > exten

Re: [asterisk-users] delay when rinigng asterisk

Tell your box to not expect Caller*ID information. You set that with usercallerid=no in /etc/asterisk/zapata.conf Since you are using the Asterisk Appliance you would have to contact Digium for support. Sydney Web Hosting wrote: > Hi All, > > I have just setup an asterisk box (AA50) and all i

Re: [asterisk-users] wait & pickup

chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8). Enrico Pasqualotto wrote: > Hi all, One question > I have set in the extensions.conf of my asterisk that all incoming call > go in the wait application because I need to not "connect" the caller > but remain in the ringing

Re: [asterisk-users] How to change http port on appliance?

You should contact Digium for support for the Asterisk Appliance. It works totally differently from other Digium products. Fidel Garcia wrote: > I just found it at : /ramfs/etc/asterisk/http.conf > > > > How do I restart the http service without affecting the phone service? > > > > Fidel

Re: [asterisk-users] Choppy audio

Make the card stop sharing it's IRQ with your IDE controller. Try moving the card to another slot. Asterisk has to send an audio packet every 20ms for VoIP calls. I believe Zaptel expects no more than a few ms of latency. If something is causing a delay, like the IDE controller locking inter

Re: [asterisk-users] GotoIfTime Function

If any docs were the cause of this (very important) misconception, maybe the docs could be reworded. Do you remember what caused you to think that context was created automatically? broadband Voice wrote: > fc7234153*CLI> dialplan show open > There is no existence of 'open' context > I was unde

Re: [asterisk-users] Trouble with PRI config

This will happen if the other side is configured the same as the Asterisk side. i.e. PRI CPU mode on both ends or PRI NET mode on both ends. This can also happen if the line is in loopback mode at the far end. Eve-Ellen Cole wrote: > > > The underscore helped, but didn't resolve the real iss

Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri

Asterisk allows you to add custom SIP headers. SER is a *very* powerful SIP proxy. I imagine you should be able to make SER translate those headers into the URI as it routes the SIP packet. Tom Browning wrote: > > To send calls into a custom SER implementation, I need to be able to add > som

Re: [asterisk-users] Canadian Whitepage Listing Capability

Joseph L. Casale wrote: > So my SIP Provider states they do not offer the service to list my numbers w/ > the Whitepages. > We phoned the Whitepages and they said we can't do it, the SIP Provider must? > > Either one/both of them is/are useless or I must switch SIP providers to one > that can g

Re: [asterisk-users] Idiot's Question

Oddly "core show function SPRINTF" works on my 1.6. SPRINTF function does not seem to be in 1.2 and I don't have any 1.4 systems. Venefax wrote: > Believe it or not, I cannot find online a single piece of documentation for > the Asterisk function SPRINTF. This example does not work, for it chang

Re: [asterisk-users] Idiot's question

This should do it, but I've not actually tested it. It is based on a line from my own dialplan. _X.,n(entrada),Set(CALLERID(num)=${IF($[${LEN(${CALLERID(num)})} = 0]?00:${CALLERID(num):0:11})}) Venefax wrote: > I have two lines in my dialplan that I wish to make it into only one, and I

Re: [asterisk-users] SIP call, updated with CID as it becomes available

Answer() is seldom the solution. Rob Hillis wrote: > Steve Totaro wrote: >> If you ever have problems with a call dropping after 30 seconds, >> Answer() is usually the cause. >> >> > > Answer is the /cause/? Or do you mean it's the solution? -- Consulting for Asterisk, Polycom, Sangoma, Di

Re: [asterisk-users] Interoffice phone setup

The external DNS server would immediately return with a not found message. Without internet access you'll have to wait for the timeouts, etc. Joseph L. Casale wrote: >> Asterisk gets very upset if it can't lookup the host name associated >> with every IP on the system, normally it would use DNS

Re: [asterisk-users] g729 open source codec and sample size

The G729 codec is neither open source, nor is it free, and the license/patent does not make an exception for "educational use". The Intel LIBRARIES are free for educational/personal use, but the license for that software says that you still need a license from the G729 patent holder before use.

Re: [asterisk-users] Fax on FXS

k-users] Fax on FXS > > On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: >> On June 9, 2008 01:34:31 pm Eric "ManxPower" Wieling wrote: >>> You should not expect FaxOverVoiceOverIPOverInternet to work well. If >>> you stick to ulaw codec for the e

Re: [asterisk-users] Interoffice phone setup

Asterisk gets very upset if it can't lookup the host name associated with every IP on the system, normally it would use DNS to do this, but since your Internet connection was down it could not do that. You should look at /etc/hosts on the Asterisk machine and make sure that each IP address of

Re: [asterisk-users] Fax on FXS

You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. John Morey wrote: > I've been thinking about something around these lines that I'd like feedback > on. What I'd like to d

Re: [asterisk-users] Asterisk not picking up incoming calls from TDM400P

Correct. The previous poster was wrong. Drew Gibson wrote: > Nope, didn't help. > > Doesn't the context declaration come *before* the channel assignment in > zapata.conf? > It's working that way in our main Asterisk server. > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN

Re: [asterisk-users] fxotune question

Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might b

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