Re: [asterisk-users] best softphone for 2012?

On 01/06/2012 05:00 PM, Tom Poe wrote: Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for incoming/outgoing calls. No video. Tom We

Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

On 1/4/2012 4:37 AM, Jayesh Labade wrote: Please help me.. Best Regards, *Jayesh Labade* e-mail: jayesh.lab...@gmail.com On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade mailto:jayesh.lab...@gmail.com>> wrote: Hello Experts, I have pasted my issue in h

Re: [asterisk-users] Speech recognition in asterisk using google voice API

On 1/4/2012 2:26 PM, Lefteris Zafiris wrote: Works beautifully. Amazing job Lefteris. Thanks. The best result I got in probability was 0.9725632 by saying, "hello". I think there is some non-phonetic logic built-in as well. I tried, "1, 2" and I got "0.86534226" in accuracy. While I tried "1, 2

Re: [asterisk-users] tcp version of toronto - osaka doesn't work

On 01/02/2012 11:30 AM, sean darcy wrote: On 01/02/2012 11:21 AM, sean darcy wrote: On 01/01/2012 11:34 PM, sean darcy wrote: I'm trying to setup a simple tcp sip connection based on the toronto osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto]

Re: [asterisk-users] tcp version of toronto - osaka doesn't work

On 01/02/2012 11:21 AM, sean darcy wrote: On 01/01/2012 11:34 PM, sean darcy wrote: I'm trying to setup a simple tcp sip connection based on the toronto osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto] type=friend transport=tcp secret=welcome co

Re: [asterisk-users] tcp version of toronto - osaka doesn't work

On 01/01/2012 11:34 PM, sean darcy wrote: I'm trying to setup a simple tcp sip connection based on the toronto osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto] type=friend transport=tcp secret=welcome context=toronto_incoming host=dynamic disallo

[asterisk-users] tcp version of toronto - osaka doesn't work

I'm trying to setup a simple tcp sip connection based on the toronto osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto] type=friend transport=tcp secret=welcome context=toronto_incoming host=dynamic disallow=all allow=ulaw sip show peer toronto * Name

[asterisk-users] can't set up tcp sip - sip connection : digest problem

Trying to set up a simple sip - sip connection over tcp. Home : 10.0.0 - Office: 1.8.8.0 Home sip.conf: register => tcp://office-going-to-home:password@/home-coming-from-office [home-coming-from-office] ; receives calls type=friend transport=tcp dtmfmode=rfc2833 disallow=all allow=ulaw

Re: [asterisk-users] odd "secret" problem

On 12/26/2011 10:05 PM, sean darcy wrote: I've now set up tcp to connect for some home-office connections. Home is 10.0.0, office is 1.8.8.0. The home sip device is home-going-to-office, office device: office-coming-from-home - home ip is 10.10.11.180 -- Called SIP/home-going-to-offic

Re: [asterisk-users] how to listen on different sip port for a device?

On 12/26/2011 08:17 PM, Jim Dickenson wrote: Why not use IAX trunk instead of SIP. This would make it very easy to talk between the two * systems. I've tried iax. I found the voice quality was better with sip. YMMV. sean -- __

[asterisk-users] odd "secret" problem

I've now set up tcp to connect for some home-office connections. Home is 10.0.0, office is 1.8.8.0. The home sip device is home-going-to-office, office device: office-coming-from-home - home ip is 10.10.11.180 -- Called SIP/home-going-to-office/166 [Dec 26 18:42:31] NOTICE[4387]: chan_sip.c:2

Re: [asterisk-users] how to listen on different sip port for a device?

On 12/26/2011 05:43 PM, Yaroslav Panych wrote: 2011/12/26 sean darcy: So how do I get * to listen to two different ports? sip.conf section [general] bindport=whatever-port-you-want Thanks, but the problem is to get more than 1 port, 5060 and (at least) one other. sean

Re: [asterisk-users] how to listen on different sip port for a device?

On 12/26/2011 10:39 AM, Kevin P. Fleming wrote: On 12/26/2011 08:55 AM, sean darcy wrote: I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to

[asterisk-users] how to listen on different sip port for a device?

I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip

[asterisk-users] android won't play wav49: how to change format

android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.co

[asterisk-users] 10-rc2: how to debug dropped calls?

I've been experiencing a number of dropped calls - both where I'm calling out and the call drops before answer, and where it's inbound and the call drops while I'm talking (usually at almost exactly 5 minutes). I'm using dahdi 2.5.0.1 with a TDM400P connected to PSTN. The console doesn't show

Re: [asterisk-users] trouble with sip connection and registration

ewall misconfig, perhaps. Or the unthinkable: your home ISP has started filtering 5060. On Nov 14, 2011, at 18:51, sean darcy wrote: I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home

[asterisk-users] trouble with sip connection and registration

I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0 on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:-- Registrat

Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card

On 11/11/2011 07:38 PM, Eric Wieling wrote: Show us /etc/asterisk/chan_dahdi.conf (and any #include'd files) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 11, 2011 5:

[asterisk-users] 10.0.0-rc1: dahdi doesn't see card

From asterisk -cv == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found -- Automatically generated pseudo channel [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'userbase' (on reload) at li

Re: [asterisk-users] 10.0.0-rc1: won't start: "empty buf size"

On 11/11/2011 05:23 PM, Kevin P. Fleming wrote: On 11/11/2011 04:19 PM, sean darcy wrote: Trying out 10.0.0-rc1. It dies starting up: == Parsing '/etc/asterisk/codecs.conf': == Found [Nov 11 17:07:05] WARNING[5078]: translate.c:1060 __ast_register_translator: empty buf size, you need

[asterisk-users] 10.0.0-rc1: won't start: "empty buf size"

Trying out 10.0.0-rc1. It dies starting up: == Parsing '/etc/asterisk/codecs.conf': == Found [Nov 11 17:07:05] WARNING[5078]: translate.c:1060 __ast_register_translator: empty buf size, you need to supply one [root@asterisk ~]# Where do I supply the "buf size" to the translator? And what s

[asterisk-users] 1.8.7.0 crashing : Can't send 10 type frames with SIP write

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s@macro-stdexten:2] Dial("SIP/teliax-0019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Usin

[asterisk-users] why doesn't "s" accept incoming call

Call from 'sip' to extension '+1xxxyyy' rejected because extension not found in context 'out'. But [out] exten => s,1,NoOp( this is the extension: ${EXTEN}) exten => s,n,Answer() exten => s,n(weasels),PlayBack(weasels-eaten-phonesys) If I set "s" to "_." it works. Shouldn't "s" wo

[asterisk-users] example sip.conf for csipsimple?

I'm trying to set up csipsimple on my Droid X. But no joy. Can't get it to register. My sip.conf: [general] tcpenable=yes [Test] transport=tcp,udp type=friend secret=mytest host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite disallow=all allow=ulaw I

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

On Mon, May 9, 2011 at 8:10 AM, Olivier wrote: > Hi, > > I would be curious to play with an Android phone with Wifi-only capability. > My plan is to install Bria on it and see if it could be used within a couple > of WiFi access points, as a high-end wireless phone. > This is first reference I've

[asterisk-users] 1.8.4-rc2: ReceiveFAX fails

On a test fax: -- Executing [s@incoming-fax:1] Set("DAHDI/4-1", "FAXFILE=/var/spool/asterisk/fax/20110415_1825") in new stack -- Executing [s@incoming-fax:2] Answer("DAHDI/4-1", "") in new stack -- Executing [s@incoming-fax:3] ReceiveFAX("DAHDI/4-1", "/var/spool/asterisk/fax/201104

[asterisk-users] OT: Have unused DID's; where to warehouse?

We have a set (about 20) of DID's that we're not using. No one calls them, and we don't need them for outgoing. I'd like to keep them for future use. We now pay $5/mo/DID to host them. Is there a way to "warehouse" them? Just put them in a bank someplace? Thanks, sean -- __

Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

On Mon, Mar 7, 2011 at 6:53 PM, Dave Platt wrote: >> I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the >> office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On >> the office side, they hear an echo of _their_ speech, not mine. >> >> The office uses sip-providers g

Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

On 03/08/2011 11:02 AM, Tim Panton wrote: On 8 Mar 2011, at 02:12, sean darcy wrote: On 03/07/2011 05:26 PM, Kevin P. Fleming wrote: On 03/07/2011 04:15 PM, sean darcy wrote: I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fi

Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

On 03/07/2011 05:26 PM, Kevin P. Fleming wrote: On 03/07/2011 04:15 PM, sean darcy wrote: I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not

[asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure

Re: [asterisk-users] ignore this test

On 03/06/2011 07:15 AM, Pezhman Lali wrote: you can not see what you send, change the config in the mailing list options On Sun, Mar 6, 2011 at 6:36 AM, sean darcy mailto:seandar...@gmail.com>> wrote: I can't seem to send anything. Let's see if this shows up. No,

[asterisk-users] imsdroid on droidX to asterisk: No matching peer found

sip.conf: [imsdroid] type=friend ;;auth=md5 ;;defaultuser=imsdroid secret=mysecret host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite callerid="IMSDroid client" disallow=all allow=ulaw I've tried with and without defaultuser and secret. sip show peer imsdroid: * Na

[asterisk-users] ignore this test

I can't seem to send anything. Let's see if this shows up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www

Re: [asterisk-users] how to use qualify times to route calls

On 03/02/2011 05:34 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Wednesday, March 02, 2011 4:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users

[asterisk-users] how to use qualify times to route calls

I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? sean -

Re: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping

On 01/18/2011 08:17 PM, Shaun Ruffell wrote: On 1/18/11 6:55 PM, sean darcy wrote: On 01/18/2011 05:27 PM, Shaun Ruffell wrote: On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping

On 01/18/2011 05:27 PM, Shaun Ruffell wrote: On 01/18/2011 04:06 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Tuesday, January 18, 2011 3:58 PM To: asterisk-users

[asterisk-users] 1.8.2: dahdi-2.4: calls dropping

Here's a call coming in over PSTN to dahdi/4, connected to a local extension dahdi/1: -- Executing [s@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in new stack .. -- Executing [s@incoming-pstn-line:6] Dial("DAHDI/4-1", "DAHDI/g0,36") in new stack -- Called g0 -- DAHD

[asterisk-users] using google for vm transcripts

I'm pretty impressed by how well (comparatively) google voice does in doing voice mail transcripts. So I'd like to have google do my local voice mail, and then email the transcript. So I set up extensions.conf: exten =>s,n,Dial(${House_Phones},36) ; this should be six rings exten =>s,n,Dial(G

Re: [asterisk-users] MeetMe -> ConfBridge: hint not working

On 12/21/2010 10:15 PM, sean darcy wrote: On 12/21/2010 10:03 PM, sean darcy wrote: On 12/21/2010 12:13 PM, Jeremy Betts wrote: What version are you running? I believe device state tracking for ConfBridge was recently added. On Tue, Dec 21, 2010 at 3:39 AM, sean darcy mailto:seandar

[asterisk-users] How to use google voice for voicemail transcription

No, I don't know how to do this. Does anybody? I'd like to take a voicemail file from asterisk (*.wav, *.gsm, *.mp3 ?) and send it to googlevoice as a voicemail, then get the transcription over gmail. I know about pygooglevoice (is it still maintained?). But I can't figure out how to dial g

Re: [asterisk-users] MeetMe -> ConfBridge: hint not working

On 12/21/2010 10:03 PM, sean darcy wrote: On 12/21/2010 12:13 PM, Jeremy Betts wrote: What version are you running? I believe device state tracking for ConfBridge was recently added. On Tue, Dec 21, 2010 at 3:39 AM, sean darcy mailto:seandar...@gmail.com>> wrote: I'm trying to m

Re: [asterisk-users] MeetMe -> ConfBridge: hint not working

On 12/21/2010 12:13 PM, Jeremy Betts wrote: What version are you running? I believe device state tracking for ConfBridge was recently added. On Tue, Dec 21, 2010 at 3:39 AM, sean darcy mailto:seandar...@gmail.com>> wrote: I'm trying to migrate from MeetMe to ConfBridge: [

[asterisk-users] MeetMe -> ConfBridge: hint not working

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBri

Re: [asterisk-users] Specifying DID for outbound calls

On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell wrote: > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese > Sent: Sunday, December 19, 2010 12:49 PM > To: Asterisk Users Mailing List - No

[asterisk-users] 1.8.1: playing imaginary sound files

-- Executing [...@incoming-pstn-line:5] VoiceMail("DAHDI/4-1", "1...@default,u") in new stack -- Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') .. But there is no /var/spool/asterisk/voicemail/default/100/unavail.gsm', indeed no u

Re: [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

On 12/10/2010 05:49 PM, Kevin P. Fleming wrote: > On 12/10/2010 04:18 PM, sean darcy wrote: >> On 12/10/2010 05:01 PM, Kevin P. Fleming wrote: >>> On 12/10/2010 03:26 PM, sean darcy wrote: >>>> On 12/10/2010 02:57 PM, Kevin P. Fleming wrote: >>>>

Re: [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

On 12/10/2010 05:01 PM, Kevin P. Fleming wrote: > On 12/10/2010 03:26 PM, sean darcy wrote: >> On 12/10/2010 02:57 PM, Kevin P. Fleming wrote: >>> On 12/10/2010 01:45 PM, sean darcy wrote: >>> >>>> This was supposedly fixed in 1.6.2 on November 22, 2010. So i

Re: [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

On 12/10/2010 02:57 PM, Kevin P. Fleming wrote: > On 12/10/2010 01:45 PM, sean darcy wrote: > >> This was supposedly fixed in 1.6.2 on November 22, 2010. So isn't the >> fix in 1.6.2.15, released 12/8? >> >> In any event, that bug has been declared fixe

Re: [asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

iginal Message - > From: "sean darcy" > To: asterisk-users@lists.digium.com > Sent: Friday, December 10, 2010 12:47:54 PM > Subject: [asterisk-users] 1.6.2.14> 1.6.2.15: blind transfer works but not > Xfer on aastra > > Upgraded from 16.2.14 to 1.6.2.15

[asterisk-users] 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: ## works for a blind transfer. XferXfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using

Re: [asterisk-users] Why are the hackers scanning for these?

On Sun, Nov 7, 2010 at 11:03 AM, Cary Fitch wrote: > Adding on more thoughts: > > Think what Google has done in Mapping the Earth, Mapping the Web, and now > working on Google Voice and Google Mail. > > Every one of those makes money either directly and/or synergistically with > other components.

Re: [asterisk-users] Why are the hackers scanning for these?

On Sun, Nov 7, 2010 at 10:00 AM, Cary Fitch wrote: > > > > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo > Sent: Sunday, November 07, 2010 8:33 AM > > To: Asterisk Users Mailing List -

Re: [asterisk-users] Under heavy attack

On 10/31/2010 11:26 AM, Joel Maslak wrote: > I suspect even munin would provide you such options. Not to mention any > more capable monitor. > > > I already have a monitor (tied into nagios, which pages me if my fraud > thresholds are exceeded), but I feel that is probably beyo

[asterisk-users] baffled by defaultuser on aastra 9133i

1.6.2.13, sip.conf: [155] type=friend context=longdistance callerid="Admin" <155> secret=test host=dynamic dtmfmode=rfc2833 allow=all defaultuser=155-trust On aastra: Basic SIP Authentication Settings Screen Name Phone Number 155 Caller ID 155 Authe

[asterisk-users] How to have failover sip interface?

My asterisk machine has 2 nic's. One nic (ETH0) is connected to a cable modem. The other nic (ETH1) is connected to an internal lan. The internal lan also has access to the internet. The cable service, Time-Warner RoadRunner, is great when up, but is not reliable. And sip connections are excelle

Re: [asterisk-users] Why such high latency on internal lan?

On 10/23/2010 12:38 PM, Doug Lytle wrote: > sean darcy wrote: >> Why are the sip latencies so high? And is it a problem? And if so, how >> do I fix it? >> > > I've noted that if I run DNS on the Asterisk sever, that my ms times > drop by almost 50% > > Doug

[asterisk-users] Why such high latency on internal lan?

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10

[asterisk-users] Why high latency on internal lan?

I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers 142/14210.10.10.42 D A 5060 OK (137 ms) 144/14410.10.10.44 D A 5060 OK (136 ms) 145/145

Re: [asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?

On Sun, Oct 17, 2010 at 3:59 AM, Olivier wrote: > mù:l;:kj,nb   hgyuè > > 2010/10/16 Frank Tarczynski >> >>  I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax >> machine.  Both are connected to a DAHDI board.  I'd like to route >> incoming PSTN fax calls to the extension of th

Re: [asterisk-users] Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?

On 10/02/2010 04:09 PM, bruce bruce wrote: > Can't I in my ip tables just accept the pap2t.dyndns.org > if that is bind to the PAP2T? do you think the > devices comes in with it's external IP rather than the dyndns domain? > > Thanks > > On Sat, Oct 2, 2010 at 3:43 PM, bru

[asterisk-users] Looking for a PSTN DTMF echo test

I'm having problems with DTMF on outgoing DAHDI. See https://issues.asterisk.org/view.php?id=18084 I've tested that the dtmf comes into * correctly. I know that it works if I use SIP outbound. But it doesn't work if I use DAHDI outbound. I'd like to figure out what the called party on outbound

Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon

On 08/12/2010 09:08 AM, unsero...@aol.com wrote: >> >> Hi all, >> >> using Asterisk 1.8 beta3 installed from scratch I am not able to > stop/start/restart Asterisk deamon with >> >> /etc/init.d/asterisk stop|start|restart >> >> It just happens nothing, no warnings, errors etc. > > Next step: st

[asterisk-users] DEBUG: Cannot find variable 'XXX' ??

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01

Re: [asterisk-users] Why asterisk down when inet server down?

On 06/18/2010 06:19 PM, Cary Fitch wrote: > > > But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No > sip. No iax. Why does the asterisk machine have to resolve any address? > > The internal phones can't even call each other, even though they have > hard ip addresses. > >> Same fo

Re: [asterisk-users] Why asterisk down when inet server down?

On 06/18/2010 05:08 PM, Hans Witvliet wrote: > On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote: > >> >> If the internet server is down, there can't be a valid DNS server >> accessible to Asterisk. The asterisk server is a caching name server, >> but obviously

Re: [asterisk-users] Why asterisk down when inet server down?

On 06/18/2010 03:09 PM, Steve Edwards wrote: > Abandoning all hope of un-top-posting... > > On Fri, 18 Jun 2010, sean darcy wrote: > > (Sean has a problem and several posters suspect it is DNS related.) > > On Fri, 18 Jun 2010, Zeeshan Zakaria wrote: > >> Did you ch

Re: [asterisk-users] Why asterisk down when inet server down?

entries from /etc/resolv file and replace them > with the IP addresses of your DNS. > > Zeeshan A Zakaria > > -- > www.ilovetovoip.com <http://www.ilovetovoip.com> > >> On 2010-06-18 1:29 PM, "sean darcy" > <mailto:seandar...@gmail.com>> wrot

Re: [asterisk-users] Why asterisk down when inet server down?

On 06/18/2010 01:19 PM, Gordon Henderson wrote: > On Fri, 18 Jun 2010, sean darcy wrote: > >> We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 >> - has a PRI connection to a T-1. Another server is the router to the >> internet. All phones in the off

Re: [asterisk-users] Why asterisk down when inet server down?

On 06/18/2010 12:57 PM, Tim Nelson wrote: > - "sean darcy" wrote: >> We have a 10.10.0.0 internal network. The asterisk server - >> 10.10.10.180 >> - has a PRI connection to a T-1. Another server is the router to the >> internet. All phones in the o

Re: [asterisk-users] Why asterisk down when inet server down?

m.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy > Sent: Friday, June 18, 2010 11:40 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Why asterisk down when inet server down? > > We have a 10.10.0.0 internal network. The asterisk ser

[asterisk-users] Why asterisk down when inet server down?

We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from

Re: [asterisk-users] How to stop intruder from registering sip?

On 06/13/2010 01:59 PM, Dave Platt wrote: >> If you leave your asterisk box open to the world with passwords like >> you deserve to be hacked.. > > Well, without making a moral judgment, I will agree that you are *going* > to be hacked if you do this! > > The O.P. seems to have made two (fairl

Re: [asterisk-users] How to stop intruder from registering sip?

On 06/13/2010 02:07 AM, dotnetdub wrote: > > The trouble with whitelisting, or using iptables to block 5060 (in fact > * is behind a router - 5060 is port forwarded) is that traveling > employees wouldn't be able to register with inbound extensions. We set > up our travelers so they

Re: [asterisk-users] How to stop intruder from registering sip?

On 06/12/2010 10:57 AM, Benoit wrote: > On 12/06/2010 15:09, sean darcy wrote: >> I decided to include the following in each sip.conf stanza that has an >> outgoing context: >> >> deny=0.0.0.0/0.0.0.0 >> permit=10.10.10.0/24 >> > If all your phones are

Re: [asterisk-users] How to stop intruder from registering sip?

sean darcy wrote: > This is a small 12 line system, internal extensions 150 - 180. I didn't > have a phone on 151. Here's the sip.conf stanza: > > ;;[151] > ;;type=friend > ;;context=longdistance > ;;callerid="Conf Room" <151> > ;;secret= >

Re: [asterisk-users] How to stop intruder from registering sip?

Fred Posner wrote: > On Jun 11, 2010, at 5:55 PM, sean darcy wrote: > >> This is a small 12 line system, internal extensions 150 - 180. I didn't >> have a phone on 151. Here's the sip.conf stanza: >> --snip-- >> There's no DISA. And then somehow

[asterisk-users] How to stop intruder from registering sip?

This is a small 12 line system, internal extensions 150 - 180. I didn't have a phone on 151. Here's the sip.conf stanza: ;;[151] ;;type=friend ;;context=longdistance ;;callerid="Conf Room" <151> ;;secret= ;;host=dynamic ;;qualify=yes ;;dtmfmode=rfc2833 ;;allow=all ;;defaultuser=151 ;;nat=yes

Re: [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel

Leif Madsen wrote: > sean darcy wrote: >> Richard Kenner wrote: >>>> Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi. >>>> If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this >>>> is my problem, ins

Re: [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel

Richard Kenner wrote: >> Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi. >> If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this >> is my problem, instead of filing. > > I reported another instance of this today and it was fixed in the SVN a few > hour

Re: [asterisk-users] 11.6.2 segfaults after dtmf on dahdi channel

Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi. If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this is my problem, instead of filing. sean Alec Davis wrote: > I filed the following bug on the 28th of May. > > 0017371: [patch] [regression] DAHDI analo

[asterisk-users] 1.6.2.8: need sip reload to reach peers.

Running 1.6.2.8 and using teliax and junction for sip providers. We also have an internal sip network for extensions. On boot both providers are shown as unspecified and UNKNOWN. The internal sip extensions are found. sip reload always finds the providers. Why doesn't asterisk find the provid

Re: [asterisk-users] working example of t38 fax w/ 1.6.2?

Ilmars Knipšis wrote: > Hello! > > I use similar setup. > Probably you need Answer() in receiving end. And wait(3) before > receiving fax. > T.38 works fine with 1.6.2. > > Ilmars. > > On 2010.05.05. 0:17, sean darcy wrote: >> On 5/4/2010 7:32 AM, Migu

[asterisk-users] Still true: only first peer matched on incoming call?

I've got two 1.6.2 asterisk boxes. I'd like to be able to set up two separate sip connections. But when I try that I get: chan_sip.c:12671 check_auth: username mismatch, have , digest has Looking around I found this in a 2007 bug report on version 1.4.4, https://issues.asterisk.org/view.php?id

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

On 5/4/2010 1:59 PM, Asterisk Development Team wrote: > The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 1.6.2.7 resolves several iss

Re: [asterisk-users] working example of t38 fax w/ 1.6.2?

On 5/4/2010 7:32 AM, Miguel Amez wrote: > App_fax? I didn't hear about that. What's that? > Could you please explain that a little bit better? > I'm experiencing some troubles with T38modem and would like to solve on > the better way. > > regards, > &

Re: [asterisk-users] working example of t38 fax w/ 1.6.2?

Miguel Amez wrote: > Hi Sean, > > Do you know about t38modem and hylafax? > There are lots of wonderfull options with both of them. > > If you need config files with both of them tell me. > > See ya > > 2010/5/2 sean darcy mailto:seandar...@gmail.com>>

[asterisk-users] working example of t38 fax w/ 1.6.2?

I can't get a test T.38 fax between 2 1.6.2 machines, using app _fax and spandsp pre17 and 20100501. The machines can't seem to get connected. send side extensions.conf: [fax-tx-test] exten=>s,1,NoOp(Context fax-tx-test) exten=>s,n,SendFAX(${FaxFile}.tif) exten=>s,n,HangUp() exten=>h,1,NoOp(FA

Re: [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]

would be nice), and just execute it like a normal command: > > [name_you_gave_it] sip > [name_you_gave_it] peer > [name_you_gave_it] whatever > > > Note: > You then could make a lot of fancy customizations to parameters of your > script, etc., and even use other tools for if n

Re: [asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]

Olle E. Johansson wrote: >> Further to Steve Edward's comment, I think things would be more >> obvious if the help system was improved slightly, for instance: >> >> If you were trying to figure out the commands dealing with peers, you >> would be able to type: >> *CLI> help peer >> No "peer" comman

Re: [asterisk-users] 1.6.2 No "soft hangup"?

Steve Edwards wrote: > On Tue, 20 Apr 2010, Jared Smith wrote: > >> On Tue, 2010-04-20 at 09:49 -0700, Steve Edwards wrote: >>> I'd like to see a more natural and intuitive interface following a "verb >>> noun" model like Oracle, MySQL, or even GDB. >> We're close to that now, and that's one of th

[asterisk-users] 1.6.2.6: can't upgrade from 1.6.1.18

I'm running 1.6.1.18 on an older ubuntu machine. I upgraded to dahi-linux-2.3.0. That went fine, and it works. But I decided to use the opportunity to upgrade to 1.6.2.6. That didn't work. configure, make menuselect, make, make install all went fine, or at least seemed to. But it hangs starting

Re: [asterisk-users] Asterisk and spandsp fax problem

Fulajtár Pál wrote: > Hi, > > I am recently updated my asterisk 1.4.1 to 1.6.2.6 because of fax > service. I have installed spandsp 0.0.5, then 0.0.6 pre 17 as well > because to have support. app_fax is available in menuselect and loaded > into my asterisk. The upgrade went without any problems.

Re: [asterisk-users] 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops

sean darcy wrote: > Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on > 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. > > -- Executing [...@fax-tx-test:3] SendFAX("SIP/side-sip-0009", > "/var/spool/asterisk/fax/20091113_1455.tif&quo

Re: [asterisk-users] Which folder for sounds?

Tzafrir Cohen wrote: > On Mon, Mar 22, 2010 at 09:38:26PM -0400, sean darcy wrote: >> 1.6.2: >> >> -- Executing [...@incoming-pstn-line:4] VoiceMail("DAHDI/4-1", >> "1...@default,u") in new stack >> -- Playing >> '/var

[asterisk-users] Which folder for sounds?

1.6.2: -- Executing [...@incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "1...@default,u") in new stack -- Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in an

[asterisk-users] 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops

Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [...@fax-tx-test:3] SendFAX("SIP/side-sip-0009", "/var/spool/asterisk/fax/20091113_1455.tif") in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178

Re: [asterisk-users] Which spandsp to use with 1.6.2?

On Wed, Mar 10, 2010 at 6:39 AM, Tzafrir Cohen wrote: > On Tue, Mar 09, 2010 at 06:20:53PM -0500, sean darcy wrote: >> Receiving a fax pstn - pstn with 1.6.2.6-rc2: >> >>      -- Executing [...@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in >> ne

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