Re: [asterisk-users] Question about voip.ms service.

I use voip.ms and have no issues using IAX and Asterisk 1.4.xx Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall Their on line config samples just work! Suggest you check your firewall and your configs, and above all post some more information If you really want to ups

[asterisk-users] Question about voip.ms service.

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I kno

[asterisk-users] Question on how many phones

Can a quad or six core server with 4 GIG RAM running asterisk 1.4 handle 1000 polycom phones. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webi

Re: [asterisk-users] Question about "null routing" calls to DIDs we don't handle

Why not setup a default catch-all route that goes to either your main line (to drive sales) or a pre-recorded message (the number you dialed is disconnected...etc), and then setup more specific pattern matches for assigned numbers? I've done this before for clients that have large blocks of did

Re: [asterisk-users] Question about "null routing" calls to DIDs we don't handle

(reposted with correct subject line, I think messing up the subject line last time prevented my question from being read. Cheers :) On Thu, Jun 2, 2011 at 12:27 PM, Jesse Thompson wrote: >> Letting a carrier use you as a carrier seems like quite a bad idea >> generally.. > > I think I would agre

Re: [asterisk-users] Question about "null routing" calls to DIDs we don't handle

On 1 Jun 2011, at 22:50, Jesse Thompson wrote: > We are managing an Asterisk installation for residential VOIP service, and we > are having a problem where all inbound calls to DIDs which are assigned to us > by our wholesaler but not yet assigned to a downstream customer get caught in > a routi

[asterisk-users] Question about "null routing" calls to DIDs we don't handle

Hello, this is Jesse with Webformix. We are managing an Asterisk installation for residential VOIP service, and we are having a problem where all inbound calls to DIDs which are assigned to us by our wholesaler but not yet assigned to a downstream customer get caught in a routing loop. ** For exa

Re: [asterisk-users] Question on AMI

Jerry Geis wrote: I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFai

[asterisk-users] Question on AMI

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout

Re: [asterisk-users] question on digium repo

On 05/16/2011 08:36 AM, Jerry Geis wrote: I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=h

[asterisk-users] question on digium repo

I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digi

Re: [asterisk-users] question on ways to activate voicemail light on polycom

Jerry Geis wrote: Is there a way in asterisk to Activate/Clear the blinking light on polycom phones I've always just dumped a msg000.txt in the voicemail directory of that phone and removed it when not needed. Under 1.4, the Polycoms act on it. Doug -- Ben Franklin quote: "Those who woul

Re: [asterisk-users] question on ways to activate voicemail light on polycom

On 11-05-06 02:56 PM, Watkins, Bradley wrote: > Yes, use the MinivmMWI application. That's how I've done it in the past as well. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? J

Re: [asterisk-users] question on ways to activate voicemail light on polycom

us immediately and then destroy it. >From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >boun...@lists.digium.com] On Behalf Of Jerry Geis >Sent: Friday, May 06, 2011 2:15 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [asterisk-users] q

Re: [asterisk-users] question on ways to activate voicemail light on polycom

On Fri, May 6, 2011 at 2:14 PM, Jerry Geis wrote: > Is there a way in asterisk to Activate/Clear the blinking light on polycom > phones > indicating VM. Either from an AGI or some way in the dialplan? > > I want to be able to control this light for from my application. > I have an AGI to do somet

[asterisk-users] question on ways to activate voicemail light on polycom

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light mysel

[asterisk-users] question on register and dnsmgr_lookup

I "thought" I has everything using IP addresses. I am not making "outside" calls this is all internal. I have a connection between two machines both running asterisk. I am using 1.8.3 and I see a lot of dnsmgr_lookup's for "mymachine". I have a register line in sip.conf that is the only place my

Re: [asterisk-users] Question About Codecs

On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote: > Hi > > I have a call into a MeetMe conference that when I do a "core show > channel" returns > > NativeFormats: 0x4 (ulaw) > WriteFormat: 0x1000 (g722) > ReadFormat: 0x1000 (g722) > > Can someone explain what the differences between N

[asterisk-users] Question About Codecs

Hi I have a call into a MeetMe conference that when I do a "core show channel" returns NativeFormats: 0x4 (ulaw) WriteFormat: 0x1000 (g722) ReadFormat: 0x1000 (g722) Can someone explain what the differences between Native, Wite and Read are? Regards Jon -- Jon Farmer Tel 07795 118140

Re: [asterisk-users] question on show channels

: Re: [asterisk-users] question on show channels On 11-03-22 10:54 AM, Jerry Geis wrote: > when I do "core show channels" I see > Channel Location State Application(Data) DAHDI/i1/8125551212- > s@default:1 Ring Wait(1) DAHDI/i1/3175551212- s@default:10 Up > BackGround(SM_ATTEND

Re: [asterisk-users] question on show channels

On 11-03-22 10:54 AM, Jerry Geis wrote: when I do "core show channels" I see Channel Location State Application(Data) DAHDI/i1/8125551212- s@default:1 Ring Wait(1) DAHDI/i1/3175551212- s@default:10 Up BackGround(SM_ATTENDANT) 2 active channels what is the "i1"?? I thought this was the actual cha

[asterisk-users] question on show channels

when I do "core show channels" I see Channel Location State Application(Data) DAHDI/i1/8125551212- s@default:1 Ring Wait(1) DAHDI/i1/3175551212- s@default:10 Up BackGround(SM_ATTENDANT) 2 active channels

Re: [asterisk-users] Question on Asterisk 1.8 and Wait()

Jerry Geis wrote: Your correct. it was a syntax change. the above works. I've always used Wait(#) in my 1.4.x dial plans. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] Question on Asterisk 1.8 and Wait()

Best guess is that syntax changed from 1.4 to 1.8. Change line to Exten => s,1,Wait(1) Danny Your correct. it was a syntax change. the above works. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digi

Re: [asterisk-users] Question on Asterisk 1.8 and Wait()

On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote: > When I switched to 1.8 from 1.4 I am getting this error > > pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension > (default, s, 1) > > http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands > Th

Re: [asterisk-users] Question on Asterisk 1.8 and Wait()

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, March 02, 2011 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Asterisk 1.8

[asterisk-users] Question on Asterisk 1.8 and Wait()

When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I

Re: [asterisk-users] Question about how traffic passes from phones

On Mon, Feb 28, 2011 at 8:49 PM, Mitch Johnson wrote: > How does a non-secure phone that is on a PBX connected to an asterisk over > a SIP trunk communicate with a secure phone connected to the Asterisk > server? > I think, although I'm not positive, that if either leg of the call doesn't offer S

[asterisk-users] Question about how traffic passes from phones

Hopefully this is a simple question. How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I like to think that the secure call terminates on the Asterisk and the non-secure call is somehow join

Re: [asterisk-users] Question about EuroBRI final 2 digits

On 2/10/11 5:54 AM, "Christian Gansberger" wrote: > Hello, > > Maybe try that: > > In your incoming isdn context: > [isdn-incoming] > exten => s,1,Set(TIMEOUT(digits)=3) > exten => s,2,WaitExten(2) > exten => s,3,Dial(SIP/operator...) > exten => 10,1,Dial(SIP/10) > exten => 20,1,Dial(SIP/20) >

Re: [asterisk-users] Question about EuroBRI final 2 digits

I reckon it's 6). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers Sent: 10 February 2011 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about Eu

Re: [asterisk-users] Question about EuroBRI final 2 digits

On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas wrote: > This sounds like a job for DISA. > http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA > helps. > If OP is using Asterisk18, perhaps we should direct him to look here?

Re: [asterisk-users] Question about EuroBRI final 2 digits

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Gansberger Sent: Thursday, February 10, 2011 5:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question

Re: [asterisk-users] Question about EuroBRI final 2 digits

Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten => s,1,Set(TIMEOUT(digits)=3) exten => s,2,WaitExten(2) exten => s,3,Dial(SIP/operator...) exten => 10,1,Dial(SIP/10) exten => 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed a

Re: [asterisk-users] Question about EuroBRI final 2 digits

: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about EuroBRI final 2 digits Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I

Re: [asterisk-users] Question about EuroBRI final 2 digits

2011/2/5 Roberto Piola > In Italy, you must enable overlapdial=yes > Is this relevant for incoming calls, as OP asked ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a li

Re: [asterisk-users] Question about EuroBRI final 2 digits

In Italy, you must enable overlapdial=yes On Thu, Feb 3, 2011 at 7:45 PM, Cassius Smith wrote: > Hello, > I have an installation in Austria; ISDN service provided by Austria Telekom. > The main number of the service is 6 digits. Incoming calls may contain 2 > additional digits, which I then use t

[asterisk-users] Question about EuroBRI final 2 digits

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when some

Re: [asterisk-users] Question About Conferencing Capabilities

Anyone else know about the holding concurrent conferences (and switching back and forth) issue ? Is it possible? And can you set up dynamic conferences that continue even when the initiator leaves? Thanks! On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA wrote: > Hi Siobhan, > > Asterisk is al

Re: [asterisk-users] Question About Conferencing Capabilities

Hi Siobhan, Asterisk is all capacity to work-on but you need to find out some way of handling conference system through WEB part , also one more thing on last point for switching between conference i am not much sure about it but i think it is possible if i will look into code implementation. reg

[asterisk-users] Question About Conferencing Capabilities

My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Frees

[asterisk-users] Question About Conferencing Capabilities

My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Frees

[asterisk-users] question on asterisk 1.8 meetme

Currently using 1.4.X and looking to JUMP to 1.8 was reading the docs and have a question. in 1.4 I could do: /usr/sbin/asterisk -rx "meetme" to see all the current meetme's. I dont see what this is now in 1.8? Thanks Jerry -- _

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

qpage -s snppserver.example.com -p lyle -f lyle test page AMARDEEP SINGH wrote: > Do you have working script? > > On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese > wrote: > > Maybe you need to read the man page for qpage. The qpage client can > send the page to

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Do you have working script? On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese wrote: > Maybe you need to read the man page for qpage. The qpage client can > send the page to an SNPP server over TCP/IP. > > Lyle > > AMARDEEP SINGH wrote: > > Our SMS-gateway is not PSTN accessible. > > > > On Thu, Jul

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: > Our SMS-gateway is not PSTN accessible. > > On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese > wrote: > > AMARDEEP SINGH wrot

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese wrote: > AMARDEEP SINGH wrote: > > Hello All, > > Scenario: > -We use asterisk as voicemail server for our cellular network. Asterisk box > is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. > -Extens

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

AMARDEEP SINGH wrote: > Hello All, > > Scenario: > -We use asterisk as voicemail server for our cellular network. > Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. > -Extensions in * are virtual, just for leaving and accessing voicemail. > > Requirement: > Asterisk to sen

[asterisk-users] Question regarding SMS(), SMSQ, SMSC

Hello All, Scenario: -We use asterisk as voicemail server for our cellular network. Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. -Extensions in * are virtual, just for leaving and accessing voicemail. Requirement: Asterisk to send SMS to cell switch(running SMSC) on

Re: [asterisk-users] question on nortel sip connection

> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Jerry Geis > Sent: Friday, June 18, 2010 9:55 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asteri

[asterisk-users] question on nortel sip connection

I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a "SIP trunk and IP address of the their server

[asterisk-users] Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing

Hi, I'm still on 1.4 and am wondering if 1.6 would fix an issue for me. Specifically, I have been given the impression that, in contrast to 1.4 which always sends packet from the default IP (if the server has multiple IPs), 1.6 sends packets back from the IP address that was used by the peers.

[asterisk-users] Question about MaxRetries in the Asterisk Outgoing folder

I'm doing some automated calling by putting .call files in the Outgoing folder of Asterisk. I'm concerned this might be a stupid question, but I'm pretty sure I've done my research well and I'm unable to come up with an answer on my own. I want to know: what happens to the .call files after the "M

[asterisk-users] Question

Ok so a while back I found an example for having a number dial multiple numbers and then whoever answers and confirms gets the call. (don't recall who the example was from, but thank you!) But Now today I've been playing with TTS and STT and came across the BackgroundDetect command. Now If I us

Re: [asterisk-users] Question about Presence and IM feature

15 jan 2010 kl. 08.23 skrev Yuji Kondo: > > > I have two questions for Asterisk feature. > > > 1. Can Asterisk support presence feature ? Asterisk is a telephony PBX and supports presence subscriptions for extension states - if a phone line is busy or not, over a few different SIP presence

[asterisk-users] Question about Presence and IM feature

Dear Team, I have two questions for Asterisk feature. 1. Can Asterisk support presence feature ? I found following information. but it is too old...? ** http://www.voip-info.org/wiki/view/Asteris

Re: [asterisk-users] Question about SIP registration

l Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Lister Sent: martedì 12 gennaio 2010 18.51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about SIP registration On

Re: [asterisk-users] Question about SIP registration

On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: > Then I have configured an account as following: > [999] > > type=friend > > username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? > Can someone explain me this kind of behaviour? Is

Re: [asterisk-users] Question about SIP registration

Instead of host=dynamic, use host=1.1.1.1, or host=1.1.1.0/255.255.255.0. Thanks, --Warren Selby On Jan 12, 2010, at 11:16 AM, Aggio Alberto wrote: Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATte

[asterisk-users] Question about SIP registration

Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind

[asterisk-users] Question about PLC of Asterisk

Hi,I want to know how to do to work PLC of Asterisk. Anyone plz help me. PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org or release note. And I see in codecs.conf, "genelicplc" setting. So I put codecs.conf in '/etc/asterisk' ,and wrote "genericplc => true". And I w

Re: [asterisk-users] question on makefile

Jerry Geis wrote: > is there no method by the configure command to --disable-FEATURE??? There is not. The Asterisk configure script is used for platform specific settings, locating libraries and header files and the like. It is not used (directly) for controlling which portions of Asterisk are bu

Re: [asterisk-users] question on makefile

> > It's a Makefile command. See: > http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554 > > great - thanks is there no method by the configure command to --disable-FEATURE??? the help says its there but doesnt seem to do anything for me. example: ./configure

Re: [asterisk-users] question on makefile

On Wednesday 06 January 2010 13:45:55 Jerry Geis wrote: > There is a line like in codes/Makefile > > $(if $(filter > codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) > > What is filter? Where is filter? > > "whereis filter" doesnt return anything > "find . | grep filter" in a

[asterisk-users] question on makefile

There is a line like in codes/Makefile $(if $(filter codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10) What is filter? Where is filter? "whereis filter" doesnt return anything "find . | grep filter" in asterisk root directory returns nothing. Thanks, Jerry _

Re: [asterisk-users] Question regarding digital card TE412p

Thanks Victor and Vinícius for the information. I will not be doing any transcoding but using some AGI scripts, I will update the status once I configure and start using them. Thanks Sandesh On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor wrote: > Hi! > > Having two TE410P with heavy load in

[asterisk-users] question on how to connect 2 boxes

Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it’s connected to E1, and its purpose to terminate calls. It will receive SIP messages

Re: [asterisk-users] Question regarding digital card TE412p

Hi! Having two TE410P with heavy load in a Pentium4 3,2GHz system running Asterisk 1.2 was no problem. It did only IVR and bridging with no transcoding though. Chris 2009/12/14 das sandesh : > Hi, > I was able to implement T122p one port PRI and was able to call out, but I > am planning to use T

Re: [asterisk-users] Question regarding digital card TE412p

The calls itselves doesn't take a lot of CPU resources, even more considering you're willing to use hardware echo cancelling. The real CPU hogs are apps like MeetMe() and AGI scripts. Those are no worse than audiotranscoding thought. You also should design the system in such way there's as few t

[asterisk-users] Question regarding digital card TE412p

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor a

Re: [asterisk-users] question on register

11 dec 2009 kl. 17.18 skrev Jerry Geis: > Where in the code does something like: > register => user[:secret[:authuse...@host[:port][/extension] > from sip.conf 1) get parsed 2) actually register. > > I tried looking in channels/chan_sip.c and don't see where that happens. > Ca

Re: [asterisk-users] question on queues

mber 13, 2009 4:20:40 PM Subject: [asterisk-users] question on queues I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup a

Re: [asterisk-users] question on queues

On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote: > I have been looking for a way from the dialplan to inquire if there are > any members in a queue. > > So what I want to do is if no users are members of a queue then I can > send the call to a given extention. > > I have the queue setup all that

[asterisk-users] question on queues

I have been looking for a way from the dialplan to inquire if there are any members in a queue. So what I want to do is if no users are members of a queue then I can send the call to a given extention. I have the queue setup all that is working. Just need to be able to send the call to a certa

[asterisk-users] question on register

Where in the code does something like: register => user[:secret[:authuse...@host[:port][/extension] from sip.conf 1) get parsed 2) actually register. I tried looking in channels/chan_sip.c and don't see where that happens. Can someone point me the right file and or function. T

Re: [asterisk-users] Question about g729

Alex Balashov wrote: > My understanding is that Asterisk will not pass through calls in codecs > for which it does not have support and/or licenses; it simply does not > advertise them in the SDP negotiation. 'support' - yes, 'licenses' - no. Asterisk supports passthrough, recording and playback

Re: [asterisk-users] Question about g729

cember 2009 01:13 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Question about g729 > > > All calls. > > Landy Landy wrote: > >>> You only need to purchase 10 licenses, if all 10 clients >>> will be maki

Re: [asterisk-users] Question about g729

Sorry for the repetition. I didn't see the other responses. -Original Message- From: Thomas Kenyon Sent: 02 December 2009 07:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about g729 Tilghman Lesher wrote: > On Tu

Re: [asterisk-users] Question about g729

ecember 2009 01:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about g729 All calls. Landy Landy wrote: >> You only need to purchase 10 licenses, if all 10 clients >> will be making calls at the same time. > > Ok. Does t

Re: [asterisk-users] Question about g729

Tilghman Lesher wrote: > On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: >> All calls. >> >> Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. >>> Ok. Does this apply only for outbound calls using a voip provider a

Re: [asterisk-users] Question about g729

On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: > All calls. > > Landy Landy wrote: > >> You only need to purchase 10 licenses, if all 10 clients > >> will be making calls at the same time. > > > > Ok. Does this apply only for outbound calls using a voip provider and/or > > applies to cal

Re: [asterisk-users] Question about g729

All calls. Landy Landy wrote: >> You only need to purchase 10 licenses, if all 10 clients >> will be making calls at the same time. > > Ok. Does this apply only for outbound calls using a voip provider and/or > applies to calls within the lan? > > > > > >

Re: [asterisk-users] Question about g729

> You only need to purchase 10 licenses, if all 10 clients > will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? ___ -- Bandwidth and Colocation Prov

Re: [asterisk-users] Question about g729

it better or correct me if Im wrong. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: 01 December 2009 23:24 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Question about

[asterisk-users] Question about g729

Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per

[asterisk-users] question about call transfer

Hi all, Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf. It shows we can use variable BLINDTRANSFER to call back the one who transfer the call. However, in my tests below. The result is not as expected. case 1: A calls B (dial(sip/B||Tt) B answers and connects to A B

Re: [asterisk-users] Question about OSLEC or HPEC with AsteriskNow

AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and asterisk with the freepbx GUI interface and it seems to be missing all the dev packages Martin On 2009-11-17, at 02:19, Olivier wrote: 2009/11/17 Martin Roy I was previously using an old computer running Asterisk 1.2 with

Re: [asterisk-users] Question about OSLEC or HPEC with AsteriskNow

2009/11/17 Martin Roy > I was previously using an old computer running Asterisk 1.2 with > zaptel. Once the CPU fried I switch to a new computer and I chose > AsteriskNow 1.5 running in 64bits to simplify the installation > process. I manage to find my way with configuring dahdi instead of > zapt

[asterisk-users] Question about OSLEC or HPEC with AsteriskNow

I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my pr

Re: [asterisk-users] Question about callerid?

OK, Now I am responding to myself, because I have figured it out (finally). It turns out it's a "feature" of asterisk (at least the older versions). This is where I found my answer: https://issues.asterisk.org/view.php?id=9678 So the solution for me was to simply rearrange my sip.conf so my

Re: [asterisk-users] Question about callerid?

Ok I am replying to myself, because I still don't have this figured out,, but I think I have more info. On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote: > > Hello again Asterisk people. > > I am running Asterisk 1.42 on an old PowerPC ibook. I have had this > deployed for several years now,

Re: [asterisk-users] Question about callerid?

On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote: > On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote: >> Hello again Asterisk people. >> >> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this >> deployed for several years now, with pretty good results. >> >> Recently

[asterisk-users] Question about callerid?

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding

[asterisk-users] question about getting instance ringing member in queue

Hi, I have a queue and 3 agents in the queue like below SIP/1001 SIP/1002 SIP/1003 When I dial the queue number, the agent start to ring. How can I get the instance ringing agent as I want to pause the agent (pausequeuemember) after the queue timeout? Any application or variable can use to g

Re: [asterisk-users] question on SIP and call manager

ion Subject: Re: [asterisk-users] question on SIP and call manager > > Here are two ways to address this > > 1. Dial(SIP/CCMMAIN&SIP/CCMSLAVE) - this tries both at once > > 2. exten => s,1,Dial(SIP/CCMMAIN,10,KkTt) >Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt) > >

Re: [asterisk-users] question on SIP and call manager

> > Here are two ways to address this > > 1. Dial(SIP/CCMMAIN&SIP/CCMSLAVE) - this tries both at once > > 2. exten => s,1,Dial(SIP/CCMMAIN,10,KkTt) >Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt) > > CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 > rings) > > Danny thats goo

Re: [asterisk-users] question on SIP and call manager

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on SIP and call manager Customer has 2 ca

[asterisk-users] question on SIP and call manager

Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a cal

Re: [asterisk-users] question on pri intense debug

Commercial Discussion Subject: [asterisk-users] question on pri intense debug Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri

Re: [asterisk-users] question on pri intense debug

On Wednesday 30 September 2009 11:54:11 Jerry Geis wrote: > > pri intense debug span > > Just pointing out that was not clear from the HELP command. > > I thought span was the span number > > not span > > Thanks for the direction. At the list level, we only provide the keywords. If you had expl

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