[Asterisk-Users] Santa Cruz, Bolivia?

I am in need of an Asterisk/Linux savvy person in Santa Cruz, Bolivia. Please contact me off-list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] doublehash patch for 1.0.1

[EMAIL PROTECTED] wrote: is there a doublehash patch for 1.0.1? o old one to res_parking.c does not apply as there is no longer res_parking.c o wiki search is useless o google only finds the problems applying old patch to 0.7 I've attached an old-school, no frills, double-hash patch

RE: [Asterisk-Users] doublehash patch for 1.0.1

[EMAIL PROTECTED] wrote: Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is,

RE: [Asterisk-Users] Disable flash hook hold?

[EMAIL PROTECTED] wrote: Currently, if I briefly press the flash hook on my phone, the caller is placed on hold. I would like for the channel to hangup if I do this instead, never placing a caller on hold (I'll be using call-parking instead). I disabled threewaycalling that is supposed to

RE: [Asterisk-Users] Am I stupid or is my card DOA.?

[EMAIL PROTECTED] wrote: On 11-Oct-2004, Alex Barnes wrote: I had/have exactly the same problem with my X100P / TDM400P dev setup. I'm also having exactly the same problem with a TDM400P I received yesterday. I'm starting to suspect that seeing it work after swapping PCI slots is a

RE: [Asterisk-Users] TDM01B Goes missing after reboot

[EMAIL PROTECTED] wrote: On Oct 12, 2004, at 7:38 PM, Ian D. Wlloughby wrote: Hi All, I have just installed a TDM01B to fix my UK callerid and echo problems. In this respect everything is wonderful, however when I reboot wcfxs fails to load due to No Device found. If I power off and on

[Asterisk-Users] Disable flash hook hold?

Trying again with a different subject... Currently, if I briefly press the flash hook on my phone, the caller is placed on hold. I would like for the channel to hangup if I do this instead, never placing a caller on hold (I'll be using call-parking instead). I disabled threewaycalling that is

RE: [Asterisk-Users] Asterisk as Internet Talk Radio PBX system

James Sutton wrote: I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back

[Asterisk-Users] TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls

Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had

RE: [Asterisk-Users] TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls

[EMAIL PROTECTED] wrote: I installed the latest CVS of everything, and we've been getting random hangups. Bruce Komito wrote: I, too, have a TDM400P with FXO cards and I am having the same problem. After further investigation, I thought that I had a bad module in the #1 position on my

RE: [Asterisk-Users] asterisk-doc Conference Call - phase 2 :)

[EMAIL PROTECTED] wrote: Broadcast with app_ices to a shoutcast server For the world to listen too :P Has anyone gotten that app_ices to accually work? I sure as hell didn't. Yes, it works. Which part are you having problems with? Can you stream something with Icecast? Which config

RE: [Asterisk-Users] Budgetone iLBC to IAX2 iLBC

[EMAIL PROTECTED] wrote: Where i can find this new firmware? Usualy i can download from http://www.grandstream.com/BETATEST/ but i only the stable version.. Thanks in advance Dimitri http://tinyurl.com/23s6m ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] cron job to reboot GS101

[EMAIL PROTECTED] wrote: I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But, you have to log in to the web interface before doing this. I've attached a php script (quick and dirty hack) that resets the specified Grandstream devices. It requires the Snoopy

RE: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.

[EMAIL PROTECTED] wrote: exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls,

RE: [Asterisk-Users] Cannot use # key to transfer calls

[EMAIL PROTECTED] wrote: I cannot use the # key to transfer a call. I have two kinds of SIP phones, Grandstream and IpDialog, and the # key cannot be used to transfer on either one. If I press the # key during a call, I hear the touchtone for it, but Asterisk does nothing. The documentation

[Asterisk-Users] IAXTel multiple registers?

With entries in sip.conf, I can route incoming SIP calls with an extension specified in the register command: register = user:[EMAIL PROTECTED]/123 The register command in iax.conf does not support specifying the extension. If I want to register multiple IAXTel accounts, how can I make them

RE: [Asterisk-Users] IAXTel multiple registers?

. [EMAIL PROTECTED] wrote: You do this with contexts attached to the [provider] section in the iax.conf file and you provide coresponding contexts and extensions in your extensions.conf file. John Barton Hodges wrote: With entries in sip.conf, I can route incoming SIP calls

RE: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?

[EMAIL PROTECTED] wrote: I have no problem transfer from one GS adaptor to another GS adaptor. /Hans-Henrik Andresen Can anyone confirm that this problem exists? The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another

[Asterisk-Users] GS HandyTone-286 Transfer Problem, can anyone confirm?

There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using

[Asterisk-Users] Zap to SIP transfer problem

I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the

RE: [Asterisk-Users] Zap to SIP transfer problem

behave the same way. [EMAIL PROTECTED] wrote: Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring

RE: [Asterisk-Users] Zap to SIP transfer problem

of the firmware on their BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never solved. Can anyone confirm this for me? I am SO SICK of dealing with HT-286 firmware bugs! [EMAIL PROTECTED] wrote: What is your ACTUAL Dial line? On Fri, 2004-03-05 at 21:19, Barton Hodges wrote: I'm using SIP

[Asterisk-Users] Stream both sides of conversation out sound card?

How feasable is it to get the Monitor app to combine the channels in pseudo-real-time and have the resulting audio stream out a soundcard? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Integrating with an existing PBX

I have successfully setup several standalone Asterisk systems and they work great. I have the opportunity to integrate Asterisk with an existing Toshiba CHSUB672A PBX. I believe that the way I should connect the systems is through a T1 interface on the Toshiba, and a T100P on the Asterisk box

[Asterisk-Users] Cause of transfer problem (GRANDSTREAM!)

It turns out that the cause of the transfer problem is the Grandstream 1.0.4.39 firmware. I was shipped a bunch of HandyTone-286 devices that contained the 1.0.4.30 firmware. This version had a bug where the phone would sometimes not ring at all. I was told by Grandstream to upgrade to the

[Asterisk-Users] Grandstream transfer solution + DTMF translation possible?

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning Unable to process inband DTMF because inband is not

[Asterisk-Users] Transfer problem

Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with Ttm flags, when I press # on the SIP phone, the pstn

[Asterisk-Users] Packet8 DTA310 Advanced Configuration

Hi, I just received a DTA310 Terminal Adapter from Packet8. The Advanced Configuration is password protected. Does anyone know the default password or algorithm necessary to get into it? Thank you, Barton ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] A solution to free line notification

Barton Hodges wrote: I've been messing around with a free line notification where an extension is dialed for a second when a line becomes available. I can't seem to get the h extension to continue when the local party hangs up. I've seen references to other people having the same problem

RE: [Asterisk-Users] Prefix the * character

[EMAIL PROTECTED] wrote: I have a smilar problem : I have a default context for an interface, where I'd like to prefix all incoming calls DID numbers (basically, the telco sends the last 4 digits dialed, I want to fully qualify my E164 number before doing extensions processing). I don't

RE: [Asterisk-Users] Prefix the * character

[EMAIL PROTECTED] wrote: On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote: The lines in a context get reordered. If you want to force the order of those lines, put the exten lines in separate contexts and include them... something like this: [some-context] include = prefix

[Asterisk-Users] Dial T option not obeyed with Grandstream BT101

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register = number:[EMAIL PROTECTED] extensions.conf: [from-sip] exten = s,1,Dial(SIP/111SIP/117) exten =

RE: [Asterisk-Users] Unable to find path from G729A to ULAW on Sipphone.com

[EMAIL PROTECTED] wrote: I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console:

RE: [Asterisk-Users] Unable to find path from G729A to ULAW onSipphone.com

[EMAIL PROTECTED] wrote: i followed what you said didint work heres what console says i cant do the 1800 call anyway -- Executing Macro(SIP/101-8376, callerid-pstn) in new stack -- Executing SetVar(SIP/101-8376, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-8376,

[Asterisk-Users] Streaming channels from Asterisk to the Internet

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to

[Asterisk-Users] Solution to dialing 800 numbers through FWD or SIPphone

Thanks to John Lodden's help, he was able to determine that the cause of my inability to dial 800 numbers through FWD or SIPphone was due to the Grandstream phone and the order of codecs in sip.conf This order breaks the 800 dialing: disallow=all allow=ulaw allow=alaw allow=g729 However, this

RE: [Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?

Barton Hodges wrote: Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXX) and can dial out to FWD users by dialing 1383FWD# However, when I

RE: [Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?

[EMAIL PROTECTED] wrote: exten = _1800XXX,1,SetCallerID(${CALLERIDNUM}) That should be = Ah yes, search and replace without forethought or inspection to include my previous email indented with for informational purposes. Alas, it does not work even with =.

[Asterisk-Users] Dialing 800 numbers through FWD or SIPphone?

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXX) and can dial out to FWD users by dialing 1383FWD# However, when I dial 18005551212 through