Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

They used to have solaris on the Digium FTP site but they seem to be gone now :( On the "free" codec site they have some complied with icc and others with gcc4 so I don't see why you can't get this working with gcc on solaris. On Jan 15, 2008 4:01 AM, Bruce McAlister <[EMAIL PROTECTED]> wrote: >

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Andrew Joakimsen wrote: > On Jan 14, 2008 7:50 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: >> > >> Anyways, buying the license is the right thing to do unless you live where >> software patent laws are not applicable. > > Totally agree. > I have bought many more licenses from asterisk than I've

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Steve Totaro wrote: > > I would suggest building it yourself > (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt > ). It is > not that difficult and ensures that it "should" be compatible with your > machine. Just a

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Andrew Joakimsen wrote: > They used to have solaris on the Digium FTP site but they seem to be gone now > :( > > On the "free" codec site they have some complied with icc and others > with gcc4 so I don't see why you can't get this working with gcc on > solaris. > If you can, be sure to submit i

[asterisk-users] Meetme recording

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ***

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Andrew Joakimsen wrote: > They used to have solaris on the Digium FTP site but they seem to be gone now > :( > > On the "free" codec site they have some complied with icc and others > with gcc4 so I don't see why you can't get this working with gcc on > solaris. > Digium do still have the Solar

Re: [asterisk-users] Asterisk 1.4.17 crashing more

Where we can see the log when this crashed coming. after that we can investigate for that particularly error. Before 1.4.17 we was using 1.2.X but we faced problem call hanged on console for one day and two day without any media and RTP. Once the call removed it comes in our billing with high

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

On Tue, Jan 15, 2008 at 09:05:35AM +, Thomas Kenyon wrote: > Andrew Joakimsen wrote: > > On Jan 14, 2008 7:50 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > >> > > > >> Anyways, buying the license is the right thing to do unless you live where > >> software patent laws are not applicable. > >

Re: [asterisk-users] Meetme recording

On Tue, 15 Jan 2008, Lees, James (UK) wrote: > > Hello, > > Is there a way to change the format from the default? > > 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format > ${MEETME_RECORDINGFORMAT}). Default filename is > meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default f

Re: [asterisk-users] Meetme recording

In article <[EMAIL PROTECTED]>, Lees, James (UK) <[EMAIL PROTECTED]> wrote: > > Hello, > > Is there a way to change the format from the default? > > 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format > ${MEETME_RECORDINGFORMAT}). Default filename is > meetme-conf-rec-${CONF

[asterisk-users] Console app

Hi all I build an Asterisk, with asterisk 1.4.16.1 source. I have notice, that the console app don't appear on CLI... Is theres some options to turn on, when I compile asterisk? Thanks... -- Gilberto Nunes Itajaí - SC ___ -- Bandwidth and Colocati

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

On Tue, Jan 15, 2008 at 11:08:33AM +, Thomas Kenyon wrote: > If there was an equivalent free codec that provided good quality audio > with such high compression and was widely supported, then I'd use it. Help make speex widely supported. Or continue to suffer with g729 and g723. --

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Tzafrir Cohen wrote: > On Tue, Jan 15, 2008 at 09:05:35AM +, Thomas Kenyon wrote: >> Andrew Joakimsen wrote: >>> On Jan 14, 2008 7:50 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: Anyways, buying the license is the right thing to do unless you live where software patent laws are not app

[asterisk-users] Meetme recording

Just set the variable ${MEETME_RECORDINGFORMAT} to the desired format and voila its done. Have fun. on Tuesday 01/15/2008 Lees, James (UK)([EMAIL PROTECTED]) wrote > > Hello, > > Is there a way to change the format from the default? > > 'r' - Record conference (records as ${MEETME_RECORD

[asterisk-users] Fax machine detect

Hi, I was recently trying out with AMD (Answering Machine Detect) to detect the status of my call if it being picked up by HUMAN or MACHINE. Just want to know if any supporting features in asterisk 1.4.11 to detect if the call enters the Fax machine. Please provide the documentation link if an

Re: [asterisk-users] CID blocking ...

J. Oquendo wrote: > Hey all, when you guys have requests from clients to block their CID > from showing through, what are others doing? I had a coworker throw in > some "Name Here"<0> garbage which none my carriers like. I don't want to > do "Private"<12345678910> so any suggestions. For SIP it

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

On Jan 15, 2008 12:57 AM, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: > On Jan 14, 2008 7:50 PM, Steve Totaro <[EMAIL PROTECTED]> > wrote: > > > > > > > I would argue that it is illegal. The main definition of illegal is " > 1. > > against law: contravening a specific law, especially a criminal l

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Tzafrir Cohen wrote: > On Tue, Jan 15, 2008 at 11:08:33AM +, Thomas Kenyon wrote: > >> If there was an equivalent free codec that provided good quality audio >> with such high compression and was widely supported, then I'd use it. > > Help make speex widely supported. Or continue to suffer w

[asterisk-users] cisco ip phne 7911G with asterisk

hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.l

[asterisk-users] SIP Reason

Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in "Decline" SIP packages, there is a header called "Reason" and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1

[asterisk-users] Playing DTMF tones down a channel

Hello, I am trying to play DTMF tones across a phone line to control the voicemail application. The voicemail app is not however detecting the tones. Does anyone have any suggestion of what I can change to help things along? I have tested playing the tones between two standard clients and the t

[asterisk-users] busy/congestion random

et("SIP/206-090a7dd8", "REALCALLERIDNUM=206") in new stack -- Executing NoOp("SIP/206-090a7dd8", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/206-090a7dd8", "1?continue") in new stack -- Goto (macro-user-cal

Re: [asterisk-users] SIP Reason

15 jan 2008 kl. 14.01 skrev Carles Pina i Estany: > > Hello, > > I'm sniffing traffic between Asterisk and a Softswitch. I see that, in > "Decline" SIP packages, there is a header called "Reason" and I would > like to access to the content of this header from Asterisk. > > How I can access to Rea

Re: [asterisk-users] SIP Reason

Hello, On Jan/15/2008, Johansson Olle E wrote: > > I'm sniffing traffic between Asterisk and a Softswitch. I see that, in > > "Decline" SIP packages, there is a header called "Reason" and I would > > like to access to the content of this header from Asterisk. > > > > How I can access to Reason h

Re: [asterisk-users] SIP Reason

Won't SIP_HEADER(reason) do that for you? e.g. exten => 1996,1,Answer exten => 1996,n,Set(sip_reason=${SIP_HEADER(reason)}) exten => 1996,n,NoOp(sip_reason) > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Johansson Olle E > Sent: 15 January

[asterisk-users] sip channel error - extension pattern matching problem

Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: ... for

Re: [asterisk-users] SIP Reason

15 jan 2008 kl. 15.39 skrev Steve Langstaff: > Won't SIP_HEADER(reason) do that for you? No, that's only works on the INVITE that opens the dialog. The reason header comes in a reply. /O > ___ -- Bandwidth and Colocation Provided by http://www.api-

[asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

A new article in my Asterisk 1.4 series cover blinking lamps on SIP business phones. Read it to learn all the new things! http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ Regards, /Olle ___ -- Bandwidth and Colocation Provided by http://

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On Tue, 2008-01-15 at 16:41 +0100, Johansson Olle E wrote: > A new article in my Asterisk 1.4 series cover blinking lamps on SIP > business phones. > Read it to learn all the new things! > > http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ Nice one Olle. Before I possibly waste my

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On 1/15/08, Johansson Olle E <[EMAIL PROTECTED]> wrote: > A new article in my Asterisk 1.4 series cover blinking lamps on SIP > business phones. > Read it to learn all the new things! > > http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ > I wonder - when this will be available from Re

Re: [asterisk-users] Playing DTMF tones down a channel

In article <[EMAIL PROTECTED]>, Lees, James (UK) <[EMAIL PROTECTED]> wrote: > > I am trying to play DTMF tones across a phone line to control the > voicemail application. The voicemail app is not however detecting the > tones. Does anyone have any suggestion of what I can change to help > things a

Re: [asterisk-users] SIP Reason

> Sent: 15 January 2008 15:23 by Johansson Olle E > > 15 jan 2008 kl. 15.39 skrev Steve Langstaff: > > > Won't SIP_HEADER(reason) do that for you? > No, that's only works on the INVITE that opens the dialog. > The reason header comes in a reply. Thanks Olle. At least no one else saw my foolish

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

Atis Lezdins wrote: > On 1/15/08, Johansson Olle E <[EMAIL PROTECTED]> wrote: >> A new article in my Asterisk 1.4 series cover blinking lamps on SIP >> business phones. >> Read it to learn all the new things! >> >> http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ >> > > I wonder - whe

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

> I wonder - when this will be available from Realtime.. Managing more > than 50 users makes static config a nightmare, and AFAIK there is no > ways how to create hints with variables/extension masks. So, it is > logical to ask for hint support in Realtime. > AFAIK hints are supported in Realtime:

Re: [asterisk-users] Asterisk 1.4 Call Recording

Are there any tricks to getting combine_wave to make? [EMAIL PROTECTED] combine_wave-0.3]# ls -al total 84 drwxr-xr-x 2 root root 4096 Jan 15 10:54 . drwxr-x--- 6 root root 4096 Jan 15 10:54 .. -rw-r--r-- 1 root root 351 Oct 6 2005 CHANGES -rw-r--r-- 1 root root 1123 Oct 6 2005 combine_wa

Re: [asterisk-users] SIP Reason

Hello, On Jan/15/2008, Steve Langstaff wrote: > > Sent: 15 January 2008 15:23 by Johansson Olle E > > > > 15 jan 2008 kl. 15.39 skrev Steve Langstaff: > > > > > Won't SIP_HEADER(reason) do that for you? > > No, that's only works on the INVITE that opens the dialog. > > The reason header comes

Re: [asterisk-users] app_voicemail for spanish

Will do AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: lunes, 14 de enero de 2008 11:48 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_voicemail for spanish No features are b

Re: [asterisk-users] Park() help, extension not heard

On Mon, 2008-01-14 at 21:02 -0800, Rob wrote: > I can place a call between two internal extensions, then on one > extension transfer the call to extension 700, and the call gets parked > on 701 but I don't hear the extension number when I do the transfer. > I can hangup and call 701 and get the cal

[asterisk-users] Fax machine detect

Hi, I was recently trying out with AMD (Answering Machine Detect) to detect the status of my call if it being picked up by HUMAN or MACHINE. Just want to know if any supporting features in asterisk 1.4.11 to detect if the call enters the Fax machine. Please provide the documentation link i

Re: [asterisk-users] Asterisk 1.4 Call Recording

Hi Mike, On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote: > Are there any tricks to getting combine_wave to make? Patch attached. Builds fine with patch on Fedora 8. Regards, Patrick diff -Naur combine_wave-0.3.orig/combine_wave.c combine_wave-0.3/combine_wave.c --- combine_wave-0.3.orig/

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On Tuesday 15 January 2008 10:51:31 CSB wrote: > > I wonder - when this will be available from Realtime.. Managing more > > than 50 users makes static config a nightmare, and AFAIK there is no > > ways how to create hints with variables/extension masks. So, it is > > logical to ask for hint support

[asterisk-users] Interrupt the swift text

Hi, I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio. My functionality is as for example i make this text to audio deliver the person called. Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to ca

Re: [asterisk-users] Asterisk 1.4 Call Recording

I'm a newb when it comes to patch. I have a combine_wave-0.3.orig and a combine_wave-0.3 directory. This is what I get: [EMAIL PROTECTED] ~]# patch < combine_wave-0.3.patch can't find file to patch at input line 4 Perhaps you should have used the -p or --strip option? The text leading up to thi

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On 1/15/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Tuesday 15 January 2008 10:51:31 CSB wrote: > > > I wonder - when this will be available from Realtime.. Managing more > > > than 50 users makes static config a nightmare, and AFAIK there is no > > > ways how to create hints with variables

Re: [asterisk-users] Asterisk 1.4 Call Recording

Never mind, I got it. I needed a -p0 - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: "Patrick" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 15, 2008 11:19 AM Subject: Re:

Re: [asterisk-users] Interrupt the swift text

On Jan 15, 2008 12:32 PM, Naveen Palani <[EMAIL PROTECTED]> wrote: > Hi, > > I am using Asterisk-1.4.11 version to make outbound calls and deliver the > swift text to audio. > > My functionality is as for example i make this text to audio deliver the > person called. > > Eg. swift -o /tmp/test.wa

Re: [asterisk-users] Interrupt the swift text

On Tuesday 15 January 2008 12:32, Naveen Palani wrote: > Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 > "Press 1 to confirm. Press 3 to cancel." Naveen, How about generating the wav files and storing them, then playing the wav's from the call tree, rather then re-gener

[asterisk-users] Heartbeat

Has anyone ever written asterisk logic to Heartbeat remote phone lines? Something that would dial out and see if a busy tone is encountered and take some sort of action? If not, any good ideas on how to do it? Obviously this would involve .call files. This e-

[asterisk-users] Record calls then send them to users voicemail

Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso

[asterisk-users] asterisk 1.4 context

I am running asterisk 1.4 with Cisco Call manager. I made a context for it of course in sip.conf when the call comes in it does not seem to be obeying the context though. Only way I could get the call to answer was to put the phone number (cut and paste the same lines here) into the default contex

Re: [asterisk-users] Interrupt the swift text

Steve/Ron, I also did that. I created a wave file and stored in /tmp directory and then use Background cmd inside macro. But it doesnt seem to work. I saw from the forum to use Background with the context parameter set to the macro context." Used it in this way, suggest me if it is wrong: exten

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On Tuesday 15 January 2008 12:13:46 Atis Lezdins wrote: > On 1/15/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > > Having a hint entry is only half the battle. The other half is in > > keeping a registry of function pointers to call when the state of the > > device changes. > > I'm not very fami

Re: [asterisk-users] Interrupt the swift text

Naveen Palani wrote: > Does anyone have any ideas i can work it out. How can i have > the Asterisk cmd Background inside macro? or how to execute the GoTo > command? I have really started to wish for 2 new standard commands - BackgroundApp and SpeechBackgroundApp to be added to Asterisk just f

Re: [asterisk-users] asterisk 1.4 context

On 14:00, Tue 15 Jan 08, Jerry Geis wrote: > I am running asterisk 1.4 with Cisco Call manager. > I made a context for it of course in sip.conf > when the call comes in it does not seem to be obeying the context though. > Only way I could get the call to answer was to put the phone number (cut > a

Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support -- Solution

Quoting Jaap Winius <[EMAIL PROTECTED]>: > Has anyone been able to get ISDN-BRI support to work reliably on > Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, > kernel, modules, versions, config files). Thanks to the support I received here I now have a working system, so

[asterisk-users] inbound Audio problems probably not NAT related?

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108

Re: [asterisk-users] Record calls then send them to users voicemail

Lookup the automon feature on features.conf . Best regards, On Jan 15, 2008 4:55 PM, Anciso, Roy <[EMAIL PROTECTED]> wrote: > Just wondering if this is possible: > > Make a call from a registered sip extension (Doesn't matter if it's > internal or external) during the call press a key sequence l

Re: [asterisk-users] Heartbeat

Have a look at the new Digium list: Asterisk-HA, I think this thread makes more sense there. Best Regards, On Jan 15, 2008 4:42 PM, Jeremy Mann <[EMAIL PROTECTED]> wrote: > Has anyone ever written asterisk logic to Heartbeat remote phone lines? > Something that would dial out and see if a busy t

[asterisk-users] Attended transfers manager or phone

Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real "simple" answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Origina

Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.

Ok, let's just agree to disagree and say that using patented software without a patent license is "wrong" What I am saying is you can be sued to the poorhouse but you won't be arrested and put in jail. On Jan 15, 2008 7:55 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > > > > > On Jan 15, 2008 12:5

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

Hello, > > I wonder - when this will be available from Realtime.. Managing more > > than 50 users makes static config a nightmare, and AFAIK there is no > > ways how to create hints with variables/extension masks. So, it is > > logical to ask for hint support in Realtime. > > > AFAIK hints are sup

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

Andrea Spadaccini wrote: > I have a small question: other than a phone (ie. SIP/something), what else can > I use as "app"? Can I handle the change via some custom code? There are a number of things that can provide "device state" in Asterisk. That includes "real" devices such as SIP endpoints,

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

Patrick wrote: > Nice one Olle. Before I possibly waste my time trying this does this > blinkety lights magic also work with SCCP phones? IIRC, this feature is currently only supported in Asterisk trunk (soon to become Asterisk 1.6). -- Russell Bryant Senior Software Engineer Open Source Team

[asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' ___ -- Bandwidth and Colocati

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

Ciao Russell, > > I have a small question: other than a phone (ie. SIP/something), what else > > can I use as "app"? Can I handle the change via some custom code? > > There are a number of things that can provide "device state" in Asterisk. > That includes "real" devices such as SIP endpoints, or

[asterisk-users] chan_mobile type=

What are the values for type for chan_mobile? headset and phone ??? I get my Treo650 to pair. hcitool scan shows the device. hcitool con comes up empty. I go into Asterisk cli. mobile search shows the device (while I am waiting for a response, I see the phone showing a connection being set up

[asterisk-users] Channel fallback

Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a w

Re: [asterisk-users] Console app

What does 'make menuselect' let you choose? Under #3, Channel Driveers, does chan_alsa have XXX through it so you can't select it? does chan_oss have XXX? This would indicate to you that the pieces of alsa or oss asterisk would need are not installed properly. Moj Gilberto Nunes wrote: > Hi

Re: [asterisk-users] CID blocking ...

On Jan 14, 2008 6:29 PM, Paul Hales <[EMAIL PROTECTED]> wrote: > > The 'setcallerpres' application is the one to use... > Only works for PRI channels (maybe plain T1) channels via Zaptel. ___ -- Bandwidth and Colocation Provided by http://www.api-digita

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

On Tuesday 15 January 2008 16:03:16 Andrea Spadaccini wrote: > Russell wrote: > > Andrea wrote: > > > I have a small question: other than a phone (ie. SIP/something), what > > > else can I use as "app"? Can I handle the change via some custom code? > > > > There are a number of things that can prov

Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Andrew Joakimsen wrote: > Anyone else have issues with T.38 where the call drops after T.38 is > attempted to be negotiated, with a message like the below? > > WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in > c= line, 'IN IP4 100101' The problem is that 100101 is neither a

Re: [asterisk-users] Channel fallback

Use the chanisavail to check that the SIP channels are clear, and set reasonable 'qualify=' settings for them PaulH On Tue, 2008-01-15 at 23:20 +0100, Jaap Winius wrote: > Hi list, > > My Asterisk v1.4 system now has two ISDN channels and two SIP > channels. The idea is to make a dialpla

Re: [asterisk-users] CID blocking ...

On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote: > On Jan 14, 2008 6:29 PM, Paul Hales <[EMAIL PROTECTED]> wrote: > > > > The 'setcallerpres' application is the one to use... > > > > Only works for PRI channels (maybe plain T1) channels via Zaptel. > Agreed entirely. PaulH

Re: [asterisk-users] Attended transfers manager or phone

Some phones have the auto-answer ability. So your phone could have two extensions, one for normal use and one for auto-answer use. Redirect or Originate, as you were, to the auto-answer extension on the phone. So the phone would already put itself offhook, and asterisk would continue and bui

Re: [asterisk-users] chan_mobile type=

Le mardi 15 janvier 2008, Robert Moskowitz a écrit : > What are the values for type for chan_mobile? > > headset and phone ??? I think that if you let #type=headset it is by default a phone. It means that asterisk will try to use the mobile as a hand's free . If you set type=headset It means that

Re: [asterisk-users] Asterisk 1.4 Call Recording

On Tue, 2008-01-15 at 12:13 -0600, Mike Hammett wrote: > I'm a newb when it comes to patch. I have a combine_wave-0.3.orig and a > combine_wave-0.3 directory. This is what I get: > > [EMAIL PROTECTED] ~]# patch < combine_wave-0.3.patch > can't find file to patch at input line 4 > Perhaps you s

Re: [asterisk-users] CID blocking ...

On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote: > On Jan 14, 2008 6:29 PM, Paul Hales <[EMAIL PROTECTED]> wrote: > > > > The 'setcallerpres' application is the one to use... > > > > Only works for PRI channels (maybe plain T1) channels via Zaptel. Just to be more complete: SetCallerPr

Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Well can you offer some explanation why T.38 faxing worked for months and then one day stopped working? Using both Linksys & Audiocodes (yuck) ATA. The first second of the fax tone is heard and then the T.38 switchover is attempted and the call drops with said error. On Jan 15, 2008 6:25 PM, M

[asterisk-users] help Unable to dial _99XXXXXXXX

Hi all This is rahul i am using asterisk 1.4.17 with degium TE120p card. I have configured the card but there is a problem coming Asterisk is dialing_98 series numbers but it is not dialing _99 showing CHANISUNAVAIL. Regards RAHUL ___ --

Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

16 jan 2008 kl. 04.43 skrev Andrew Joakimsen: > Well can you offer some explanation why T.38 faxing worked for months > and then one day stopped working? You are asking the wrong forum. Your device is clearly sending a bad SDP. Ask the vendor of that device. /O > > > Using both Linksys & Audioc

Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

16 jan 2008 kl. 00.03 skrev Tilghman Lesher: > On Tuesday 15 January 2008 16:03:16 Andrea Spadaccini wrote: >> Russell wrote: >>> Andrea wrote: I have a small question: other than a phone (ie. SIP/something), what else can I use as "app"? Can I handle the change via some custom