Re: [asterisk-users] TE121 on Asterisk

Oguzhan Kayhan schreef: Oguzhan Kayhan wrote: I want to change it to E1 instead of T1. here comes the problem. If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 Doug Yes, I just noticed the jumper on the card. Thanks a lot.

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Hello Users, Is anyone aware about a solution to call incoming number and dictate the files by using Dictate feature of Asterisk used for Medical Transcription industry. Looking forward for help. Thanks, Amit ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher wrote: > On Wednesday 25 February 2009 11:19:08 sean darcy wrote: >> Tilghman Lesher wrote: >>> On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: > On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: >> Barry L. Kline wrote: >>

Re: [asterisk-users] DID's in a specific rate center

On Wed, 25 Feb 2009, M Hulber wrote: > If you don't and you just need outbound channels you can buy one (or > more) DIDs... Wouldn't it be more accurate to say "rent" rather than "buy?" Thanks in advance, Steve Edwards

Re: [asterisk-users] DID's in a specific rate center

There are sites which will show you the ratecenter and the company. Try the following to see all NPANXX's in that ratecenter: To find the ratecenter: Tr

Re: [asterisk-users] TE121 on Asterisk

>> Oguzhan Kayhan wrote: >>> I want to change it to E1 instead of T1. >>> here comes the problem. >>> >> >> If it's anything like the older cards, there is a jumper on the card >> that sets it to T1/E1 >> >> Doug >> > Yes, > I just noticed the jumper on the card. > Thanks a lot. > > Yes i changed t

[asterisk-users] codec_dahdi and Asterisk 1.6.0.6

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would i

[asterisk-users] asterisk 1.6.0.5 and IM

hi all, i have 2 x-lite version 3.0 softphones configured on extension 9000 and 9005. i have one call the other and then try and send an IM between them using the x-lite IM facility. the asterisk console shows the message... WARNING[27193]: chan_sip.c:11866 receive_message: Received message t

Re: [asterisk-users] DTMF Forwarking Problems.

Check features.conf - all the codes are set in that file. PaulH Catalin S. wrote: > Hello ppl, > I have a problem with my asterisk when i want to call some destination > through my peers and I must enter DTMF digits to select some > extension/conference number or password to access some featur

[asterisk-users] Problems with Outbound Calls

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point wh

[asterisk-users] DTMF Forwarking Problems.

Hello ppl, I have a problem with my asterisk when i want to call some destination through my peers and I must enter DTMF digits to select some extension/conference number or password to access some features.Every numbers is accepted but when i must press # key my asterisk interpret it like transfer

Re: [asterisk-users] AGI problem using mono (.Net)

On Wed, 25 Feb 2009, Steve Edwards wrote: > The AGI interface (is that redundant?) can be summarized as: > > 1) Asterisk sends a bunch of cruft (the AGI environment variables) to your > program's STDIN. 1a) Your program must read all of the AGI environment variables. > 2) Your program sends a re

Re: [asterisk-users] AGI problem using mono (.Net)

Suggest, Use .net to do an web services and use curl+agi scripts to integrate your solutions. Regards, Luis Morales On Wed, Feb 25, 2009 at 6:37 PM, Douglas Mortensen wrote: > Hello. > > I have a software developer creating a .Net / mono program to use as an > AGI script. We are having proble

Re: [asterisk-users] Realtime database function help

You can use the MYSQL function to just use an insert or update statement in your dialplan. Look at my example below. Instead of using exten => s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\ blacklist\ where\ callerid=${ARG1} and blockenabled = 1) You could use: exten => s,2,MYSQL(Q

Re: [asterisk-users] Call from '6000' to extension rejected because extension not found

Please read this book: http://downloads.oreilly.com/books/9780596510480.pdf PaulH Chuck Coleman wrote: > > Call from '6000' to extension 'xx' rejected because extension > not found. > > > > ___

Re: [asterisk-users] AGI problem using mono (.Net)

On Wed, 25 Feb 2009, Douglas Mortensen wrote: > I have a software developer creating a .Net / mono program to use as an > AGI script. We are having problems getting it to stream files. From what > we can tell, it is talking to asterisk correctly when called from the > dial plan. Its stderr outp

Re: [asterisk-users] AGI problem using mono (.Net)

Douglas Mortensen wrote: > > I have a software developer creating a .Net / mono program to use as an > AGI script. We are having problems getting it to stream files. From what > we can tell, it is talking to asterisk correctly when called from the > dial plan. Its stderr output goes to the asteris

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher wrote: . > ... but I absolutely > defend fixing this bug in Gosub, given that I'm the designer of it, and it was > never supposed to fail into the "i" extension. > Wow. sean ___ -- Bandwidth and Colocation Provided by

[asterisk-users] AGI problem using mono (.Net)

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk does

Re: [asterisk-users] DID's in a specific rate center

Vikas, www.didforsale.com can get you the DIDs, please contact me off list. Jai Rangi jpra...@didforsale.com On Wed, Feb 25, 2009 at 1:35 PM, Vikas wrote: > > Since it's not clear from this thread of conversation, do you need 100 > > unique DIDs? > > I apologize for not being more clear. I need

Re: [asterisk-users] SheevaPlug Development Kit

On Wednesday 25 February 2009 14:59:02 Kristian Kielhofner wrote: > Hello everyone, > > I just ordered one of these: > > http://www.marvell.com/products/embedded_processors/developer/kirkwood/shee >vaplug.jsp > > Just over $110 with shipping but they are expecting the price to > come down quite

Re: [asterisk-users] Call from '6000' to extension rejected because extension not found

Is this a question? Haha. "Computer won't doesn't turn on. Got blck scrn." From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman Sent: Wednesday, February 25, 2009 3:11 PM To: asterisk-users@lists.digium.com Subject: [asteris

Re: [asterisk-users] DID's in a specific rate center

Any idea what legal statues setting caller-id fraudulently falls under? Is there a federal law you can reference? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber Sent: Wednesday, February 25, 2009 4:13

Re: [asterisk-users] DID's in a specific rate center

> Since it's not clear from this thread of conversation, do you need 100 > unique DIDs? I apologize for not being more clear. I need 100 DID's. I already have channels which allow me to set the outgoing caller id. Depending on which extension is making the call I will be sending out the unique cal

Re: [asterisk-users] SheevaPlug Development Kit

Brent Vrieze wrote: > Yes please let us know how it works out. I have several projects in the > works that this might work for. > sounds like a direct competitor of the nslu2's - The community following there is phenominal, but its nice to have some choice of platform as well. > David fire w

Re: [asterisk-users] SheevaPlug Development Kit

Yes please let us know how it works out. I have several projects in the works that this might work for. David fire wrote: > please keep us informed about it. > David > > 2009/2/25 Kristian Kielhofner > > > Hello everyone, > > I just ordered one of

Re: [asterisk-users] SheevaPlug Development Kit

please keep us informed about it. David 2009/2/25 Kristian Kielhofner > Hello everyone, > > I just ordered one of these: > > > http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp > > Just over $110 with shipping but they are expecting the price to > come down

Re: [asterisk-users] DID's in a specific rate center

Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can t

[asterisk-users] SheevaPlug Development Kit

Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit eth

[asterisk-users] Realtime database function help

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the val

Re: [asterisk-users] DID's in a specific rate center

Just contact one of the providers mentioned in this forum, such as didvv.com, broadband.com or numerous others. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 2:20 PM To

Re: [asterisk-users] DID's in a specific rate center

Spoofing the caller id is not an option for me. I am wondering how do I go about buying the DID's Thanks, Vikas On Wed, Feb 25, 2009 at 2:16 PM, Danny Nicholas wrote: > So they are going to (eventually) make a legitimate (in some cases) practice > Illegal because of spammers.  Another blow for

[asterisk-users] Congestion Tone

Hello! I’ve connected an avaya PABX with an asterisk box through h323, all calls from Avaya are sended to the asterisk. What I need is send to the AVAYA PABX a congestion tone when Zap channels are full. How I do it?Thanks for any idea! Cheers! Gustavo A. González Dto. de Infraestructura Despe

Re: [asterisk-users] Call from '6000' to extension rejected becauseextension not found

Dialplan problem, Chuck. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman Sent: Wednesday, February 25, 2009 2:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call from '6000' to extension re

Re: [asterisk-users] DID's in a specific rate center

So they are going to (eventually) make a legitimate (in some cases) practice Illegal because of spammers. Another blow for Libertarianism in the U.S. ! Don't know how this effects overseas readers. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists

[asterisk-users] Call from '6000' to extension rejected because extension not found

Call from '6000' to extension 'xx' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mai

Re: [asterisk-users] DID's in a specific rate center

http://en.wikipedia.org/wiki/Caller_ID_spoofing Danny Nicholas wrote: Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Why not expand the usage of the i extension? If not in 1.6.0, then some later 1.6. Call it a feature enhancement. Tilghman Lesher wrote: > On Wednesday 25 February 2009 11:19:08 sean darcy wrote: >> Tilghman Lesher wrote: >>> On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Til

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

On Wednesday 25 February 2009 11:19:08 sean darcy wrote: > Tilghman Lesher wrote: > > On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: > >> Tilghman Lesher schrieb: > >>> On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: > Barry L. Kline wrote: > > that is supposed to

[asterisk-users] Patton 5.3. How to get incoming calls ?

Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- -- Patton 4638 -- Asterisk -- -- I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in

Re: [asterisk-users] DID's in a specific rate center

If you have 1 real DID that you spoof from, the user will call back the real DID. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 1:06 PM To: Asterisk Users Mailing List

Re: [asterisk-users] DID's in a specific rate center

Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wed

Re: [asterisk-users] DID's in a specific rate center

Danny Nicholas wrote: > If you're using them outgoing only, you should consider "spoofing" the > number (IE calling using XXX-XXX- and presenting as 916-854-). This > would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. > > You do know that that's illegal, right? _

Re: [asterisk-users] DID's in a specific rate center

>you should consider "spoofing" the number (IE calling using XXX-XXX- and >presenting as 916-854-). But if I spoof the DID the person receiving the call will not be able to get back to me. So I do not think that is going to work for me. Vikas On Wed, Feb 25, 2009 at 12:56 PM, Danny Nic

Re: [asterisk-users] cannot allocate memory

On Wed, 25 Feb 2009, wassim Darwish wrote: > i have a hosted server with asterisk and a2billing as a billing > plattform, when i am trying to enter the server remotely by ssh, memory > error message displayed: -bash: fork: Cannot allocate memory This is not an Asterisk error message. > i have

[asterisk-users] CDR - Asterisk-Stat and PHP5

Hi all, I don't know if its the right place to ask, but... Does any one have the asterisk-stat-v2 running with PHP5? Tks! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you a

Re: [asterisk-users] DID's in a specific rate center

If you're using them outgoing only, you should consider "spoofing" the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. -Original Message- From: asterisk-users-boun...@lists.digium.com [ma

[asterisk-users] DID's in a specific rate center

I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ast

[asterisk-users] cannot allocate memory

Hi: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1

Re: [asterisk-users] Multiple SIPGate accounts.

Thanks Klaus. Putting both in the same context solved my issue! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher schrieb: > On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: >> Tilghman Lesher schrieb: >>> On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: > that is supposed to gosub into the incoming extension at priority 1. > Versions b

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

On Wed, 2009-02-25 at 11:37 -0500, M Hulber wrote: > So I'm thinking, would this work if I had a sip_.conf as well as a > sip_.conf? What the relationship between the LINEs in the > sip_.cfg and the Reg on the phone? What's the relationship > between the AUTH and the LINEn_AUTH? Th

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher wrote: > On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: >> Tilghman Lesher schrieb: >>> On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: > that is supposed to gosub into the incoming extension at priority 1. > Versions befor

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: > Tilghman Lesher schrieb: > > On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: > >> Barry L. Kline wrote: > >>> that is supposed to gosub into the incoming extension at priority 1. > >>> Versions before 1.6.0.6 would drop into

Re: [asterisk-users] Aastra phones

> > One thing which I can't figure out, although it certainly looks simple, > > is to update the firmware though FTP (not TFTP). I have set the ftp > > provisioning server in the Aastra phone, and put the firmware file > > 9143i.st in the root folder where the login/password pair ends up. > > Ever

Re: [asterisk-users] SIP_CODEC variable

Thanks, I took it for granted that the phones did support gsm...silly me. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Jared Smith > Sent: Wednesday, February 25, 2009 9:15 > To: Asterisk Users M

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

Bob, Ok, that's the route I ended up taking where all lines are the same user. I put the AUTH an LINEn_AUTH in the phone instead. I wanted to be able to set up so that each line is a different peer like below: sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CAL

Re: [asterisk-users] dahdi wcb4xxp and fax

On Wed, Feb 25, 2009 at 5:28 PM, Steve Underwood wrote: > Lee Howard wrote: > > Make sure that you're using the latest mISDN drivers. > > > Even the latest mISDN gives variable results. Some people say its OK. > Some people say its hopeless. It probably varies with the machine its > running in. >

Re: [asterisk-users] dahdi wcb4xxp and fax

Lee Howard wrote: > stoffell wrote: > >> I wanted to switch from my current setup (mISDN) to the native dahdi >> with b410p support (wcb4xp). All works fine for normal phone calls but >> not for faxing. Faxes are distorted, if arriving at all, and hylafax >> logs the usual bad stuff (HDLC fra

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

I agree with the comments on the intended target market for this phone. In defense of Polycom, if your TFTP server is external you could connect to a remote access point by setting up WEP/WPA fairly easily from Starbucks or wherever you are. If it requires web authentication to get through th

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote: > Aha! Mind posting that config? My sip_allusers.cfg looks like this: CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.8.1:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PR

Re: [asterisk-users] dahdi wcb4xxp and fax

stoffell wrote: > I wanted to switch from my current setup (mISDN) to the native dahdi > with b410p support (wcb4xp). All works fine for normal phone calls but > not for faxing. Faxes are distorted, if arriving at all, and hylafax > logs the usual bad stuff (HDLC frame not byte-oriented.) Make

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher schrieb: > On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: >> Barry L. Kline wrote: >>> that is supposed to gosub into the incoming extension at priority 1. >>> Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the >>> requested extension wasn't prese

Re: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA

Frank Bulk wrote: > It all has to do with interconnection agreements with the ILEC and if the > reseller has numbering resources in the requested area. Looks like > BroadVoice does have all those elements taken care of. > why not just lie about your address ? that would seem like the obvious

Re: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA

It all has to do with interconnection agreements with the ILEC and if the reseller has numbering resources in the requested area. Looks like BroadVoice does have all those elements taken care of. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-user

Re: [asterisk-users] SIP_CODEC variable

2009/2/25 Jeff LaCoursiere > > On Wed, 25 Feb 2009, Jared Smith wrote: > > > On Wed, 2009-02-25 at 07:54 -0500, Mike wrote: > >> I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the > >> appropriate Page command call. But I get this in th CLI: > > > >> NOTICE[4764]: chan_sip.c:3

Re: [asterisk-users] TE121 on Asterisk

> Oguzhan Kayhan wrote: >> I want to change it to E1 instead of T1. >> here comes the problem. >> > > If it's anything like the older cards, there is a jumper on the card > that sets it to T1/E1 > > Doug > Yes, I just noticed the jumper on the card. Thanks a lot. > > -- > > Ben Franklin quote:

Re: [asterisk-users] TE121 on Asterisk

> E-1s are 30 channels with D-Channel on 16. > > Ok, so i replaced the channels as D-chan 16 And now i get the following error. This card is suppose to be both e1-t1 as i understand...Or did i receive a card with only T1 support?? How will i configure it to work with e1? dahdi_cfg -vvv DAHDI

[asterisk-users] dahdi wcb4xxp and fax

Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium

Re: [asterisk-users] TE121 on Asterisk

On Wed, 2009-02-25 at 17:00 +0200, Oguzhan Kayhan wrote: > I want to change it to E1 instead of T1. To change it from T1 mode to E1 mode, you need to move the jumper on the card. (If you don't have physical access to the card, you can also override the jumper with a parameter to the kernel module

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jared Smith wrote: > While I personally believe it's a bug, it has been in Asterisk for a > very long time, and I know from teaching Asterisk training classes that > there are *many* *many* people abusing this in their dialplans. I'd be > quite hesita

Re: [asterisk-users] TE121 on Asterisk

Oguzhan Kayhan wrote: > I want to change it to E1 instead of T1. > here comes the problem. > If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

On Wed, 25 Feb 2009, Bob Pierce wrote: > Mark, > > Are you still having trouble with your 8002? I had a lot of trouble with > mine initially, but after playing with it for about 8 hours I figured it > out. Now it works great all around our office. Our NOC technician loves > it! > > There is a pro

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

On Tue, 24 Feb 2009, Michael Graves wrote: > It seems to me that based upon your comments you miss the point of the > product. It's design targets large commercial concerns, school > campuses, corporate parks, etc...not making free calls from Starbucks. Completely right. I assumed it was a gene

Re: [asterisk-users] TE121 on Asterisk

E-1s are 30 channels with D-Channel on 16. > to make it work as E1, if i write a new span like span=1,1,0,ccs,hdb3,crc4 > i got the following error when i type dahdi_cfg > dahdi_cfg - > DAHDI Tools Version - 2.1.0.2 > > DAHDI Version: 2.1.0.4 > Echo Canceller(s): MG2 > Configuration > ==

[asterisk-users] TE121 on Asterisk

Hello, I just bought a TE121 T1/E1 card, and now trying to install it on a 1.4.23.1 asterisk with dahdi 2.1.0.4 Actually first everything went on well and i managed to see my card on dahdi. Here's the output: #asterisk# dahdi_hardware pci::04:08.0 wcte12xp+d161:8000 Wildcard TE121 and

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

Mark, Are you still having trouble with your 8002? I had a lot of trouble with mine initially, but after playing with it for about 8 hours I figured it out. Now it works great all around our office. Our NOC technician loves it! There is a problem with the sample configs that Polycom publishes. I

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

On Tue, 2009-02-24 at 16:58 -0600, Tilghman Lesher wrote: > If Goto behaves that way, that's a bug. As stated in a prior email, the > "i" extension should only be implicitly invoked when waiting for a new > extension and the typed extension does not match anything. While I personally believe it's

Re: [asterisk-users] Stuck Parked Calls?

BTW, hate to reply to myself, but here is what "core show channels" shows for the stuck call: SIP/2754-0849ce682...@parkreturn:1Up (None) Also, below is the core show channel on the SIP channel: -- General -- Name: SIP/2754-0849ce68 Type: SIP UniqueID:

[asterisk-users] Stuck Parked Calls?

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind

Re: [asterisk-users] SIP_CODEC variable

On Wed, 25 Feb 2009, Jared Smith wrote: > On Wed, 2009-02-25 at 07:54 -0500, Mike wrote: >> I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the >> appropriate Page command call. But I get this in th CLI: > >> NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring >> ${

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Tilghman Lesher wrote: > On Tuesday 24 February 2009 13:44:25 Barry L. Kline wrote: >> Here's one that may be of interest to any upgraders. If you rely on the >> behavior of gosub you may want to make note of this change. >> >> I have an incoming call context: >> >> exten => _,n,GoSub(incoming

Re: [asterisk-users] SIP_CODEC variable

On Wed, 2009-02-25 at 07:54 -0500, Mike wrote: > I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the > appropriate Page command call. But I get this in th CLI: > NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring > ${SIP_CODEC} variable because it is not shared by b

Re: [asterisk-users] GSM codec is a good choice ???

On Wed, 25 Feb 2009 15:46:09 +0200, Tzafrir Cohen wrote: >On Wed, Feb 25, 2009 at 07:25:10AM -0600, Michael Graves wrote: > >> The trouble with Speex is that it has extremely limited support in >> hardware. I've yet to see a high quality IP phone that supports Speex >> directly. > >OTOH, it's well

Re: [asterisk-users] GSM codec is a good choice ???

On Wed, Feb 25, 2009 at 07:25:10AM -0600, Michael Graves wrote: > The trouble with Speex is that it has extremely limited support in > hardware. I've yet to see a high quality IP phone that supports Speex > directly. OTOH, it's well supported in soft phones. -- Tzafrir Cohen icq#

Re: [asterisk-users] GSM codec is a good choice ???

On Wed, 25 Feb 2009 09:33:42 +0200, Tzafrir Cohen wrote: >On Tue, Feb 24, 2009 at 11:16:51PM -0200, David fire wrote: >> out there is a free for educational and no commercial G729 lib for asterisk >> you can use it to test in a non-comercial system. > >For personal use? Maybe. For educational use:

Re: [asterisk-users] GSM codec is a good choice ???

2009/2/25 Alejandro Cabrera Obed > But in my case, I don't need trascoding because every chanel is in GSM > and voicemail has gsm sound files. > > And for the moment, my Asterisk is not connected to the PSTN, so there > is no trascoding gsm-to-PCM or to analog. > > So I think gsm is a good choice

Re: [asterisk-users] multiple asterisks in a server

yes, you need to make sure bindaddr is set correctly in iax.conf, sip.conf, dundi.conf, manager.conf and any other files that might include bindaddr for BOTH instances of asterisk, you can't allow one to bind to all ip's and the other just to bind to one - it won't work. 2009/2/25 Rilawich Ango

[asterisk-users] SIP_CODEC variable

Hi, I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec being used. I have exclusively Polycom phones for this test, and basically I want all communications to use g729 (preferred codec), except for pagine 20 phones (which busts my g729 license count). In that case I wan

[asterisk-users] usegmtime=yes for cdr_custom

Hi! I have set usegmtime=yes in cdr.conf, but unfortunately this is only for cdr-csv, not for cdr-custom. AFAIS there is no such option for cdr_custom.conf. Is there any workaround to get GMT timestamps in cdr-custom too? thanks klaus ___ -- Bandwid

Re: [asterisk-users] strange text message:)

I don't know what is MWI Message. All I know is that i can find these messages in my SMS inbox and has the sender voicem...@mydomain.xxx On 2/24/09, OCG Technical Support wrote: > Are you sure this is not just a standard SIP MWI message? > > > -Original Message- > From: asterisk-users-b

Re: [asterisk-users] GSM codec is a good choice ???

But in my case, I don't need trascoding because every chanel is in GSM and voicemail has gsm sound files. And for the moment, my Asterisk is not connected to the PSTN, so there is no trascoding gsm-to-PCM or to analog. So I think gsm is a good choice for my scenario, do you ??? Thanks a lot !!!

Re: [asterisk-users] switchtype QSIG and Asterisk implementation

--- On Wed, 2/25/09, Olivier wrote: > So if, euroisdn support has been dropped in the PBX > you're trying to > inconnect with, that may come from the company that > installed this PBX and > deliberately choosed to drop euroisdn feature, for a > reason. I "trust" this company and they are sayin

Re: [asterisk-users] trunk to trunk

Glad I could help!! :-D Leonja Cerebro wrote: To Robert Broyles, Thank you very much, it is very helpful information. Regards, Leonid 2009/2/18 Robert Broyles mailto:rob...@poornam.com>> Hi, You might want to check out this tutorial: http://hostseries.com/connecting-to-asterisk-

Re: [asterisk-users] trunk to trunk

To Robert Broyles,Thank you very much, it is very helpful information. Regards, Leonid 2009/2/18 Robert Broyles > Hi, > > You might want to check out this tutorial: > http://hostseries.com/connecting-to-asterisk-servers-via-sip/ > > It's a good place to start. > > -- > Regards, > Robert Broyle

Re: [asterisk-users] Asterisk with Internet connectivity

On 25 Feb 2009, at 10:38, Klaus Darilion wrote: > I have a setup with Asterisk in front of a PBX connected with ISDN to > the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing > ENUM for outgoing calls and allows incoming calls per SIP. > > Recently the IP connectivity for this loca

Re: [asterisk-users] Asterisk with Internet connectivity

2009/2/25 Klaus Darilion > Hi! > > I have a setup with Asterisk in front of a PBX connected with ISDN to > the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing > ENUM for outgoing calls and allows incoming calls per SIP. > > Recently the IP connectivity for this location was down

Re: [asterisk-users] Asterisk with Internet connectivity

Klaus Darilion a écrit : > Hi! > Hallo > I have a setup with Asterisk in front of a PBX connected with ISDN to > the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing > ENUM for outgoing calls and allows incoming calls per SIP. > > Recently the IP connectivity for this location

Re: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA

> > I am wondering how can I get the bandwidth.com service, > > Why would you persist with a company who can't service your needs when > there are others who will? +1 This industry is full of companies staffed by morons who don't give a s*. Then these companies go bust... and the idiot owners w

Re: [asterisk-users] switchtype QSIG and Asterisk implementation

Hi, On Wed, Feb 25, 2009 at 10:02 AM, Vieri wrote: > Is Asterisk "fully QSIG-compliant"? > > I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. > Zaptel versions are 1.2.26 and 1.4.11. That is a good question. I had the same dilemma here. Finally I am using my OXE via Q.SIG but d

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