[asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

I have an asterisk server at home. I'm looking to replace my internal phones with sip cordless (DECT) phones. I'm now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45). The Siemens has a feature were I can also use a PS

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

On 12/11/2012 04:37 PM, Roy Abshire wrote: That is true about the A580. I don't like the interface much to check messages. Besides that every time I go to dial a number...it always uses the first digit pressed to go into phone mode..so I have to press the first digit twice... I would test othe

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

On 12/11/2012 10:12 PM, Mitul Limbani wrote: snom m9 dect ip But it's 2-3 x the price! sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] how to join calls - not barge?

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A

Re: [asterisk-users] how to join calls - not barge?

On 02/12/2013 05:37 PM, Rusty Newton wrote: Original Message - From: "sean darcy" Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A <-> B channel? Or is there a more straight forward way to do this? The Asterisk Defi

[asterisk-users] motif - gv not working today?

I had motif working two days ago but now: Executing [1171@internal:1] Dial("DAHDI/1-1", "Motif/1171") in new stack [Feb 12 20:56:18] ERROR[7794][C-0001]: chan_motif.c:1762 jingle_request: Unable to determine endpoint name and target. motif.conf: [11XX](!) transport=google-v1 disallow=all

Re: [asterisk-users] how to join calls - not barge?

On 02/13/2013 09:39 AM, Matthew Jordan wrote: On 02/12/2013 06:48 PM, sean darcy wrote: On 02/12/2013 05:37 PM, Rusty Newton wrote: Original Message - From: "sean darcy" Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A <

[asterisk-users] 11.3: how to hang up on google voice

Some calls I get from google voice, I just send myself an email about the call and want to hangup. But I can't seem to make gv know I've hung up. extensions.conf: same => n,GoToIf($["${CALLERID(num)}"="office"]?email) . same => n(email),System(/usr/local/bin/emailme) s

Re: [asterisk-users] 11.3: how to hang up on google voice

On 03/07/2013 09:48 AM, Joshua Colp wrote: sean darcy wrote: Some calls I get from google voice, I just send myself an email about the call and want to hangup. But I can't seem to make gv know I've hung up. extensions.conf: same => n,GoToIf($["${CALLERID(num)

[asterisk-users] set google voice callerid as Unknown/Unavailable ?

I know you that GV won't respect CALLERID from motif, but is there a way have GV show Unknown on outgoing calls. I don't want to have people think my GV number is really my number. sean -- _ -- Bandwidth and Colocation Provid

[asterisk-users] 11.4.-rc1: new segfault in iksemel ??

I rebooted our server Fedora 17 today, and now asterisk won't start; asterisk[1063]: segfault at 0 ip 7f117aee122d sp 7fffbc398990 error 4 in libiksemel.so.3.1.1[7f117aed8000+d000] iksemel is required for motif and xmpp. I downloaded the iksemel source and rebuilt. No luck. Any help

[asterisk-users] 11.4: no incoming gv/xmpp

I've set up google voice to chat with me: Forwards calls to: @gmail.com and xmpp: [general] debug=no; Enable debugging (disabled by default). autoprune=yes ; Auto remove users from buddy list. Depending on your

[asterisk-users] 11.4: motif can only handle one channel at a time?

I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

On 05/16/2013 09:41 AM, sean darcy wrote: I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean More: Two different motif sections. Two different xmpp sections. xmpp

Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

On 05/16/2013 10:07 AM, sean darcy wrote: On 05/16/2013 09:41 AM, sean darcy wrote: I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean More: Two different motif

[asterisk-users] how to send dtmf after pause ?

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551...@sip.com,60,rD(12345#) The dtmf is sent too soon. I tried

Re: [asterisk-users] how to send dtmf after pause ?

On 06/07/2013 01:17 PM, Yves A. wrote: This would be possible with an agi... the agi can wait for silence or 10 seconds, as u like and then play the dtmf tones and bridge the call to your extension afterwards. yves Am 07.06.2013 17:51, schrieb Sean Darcy: I'm trying to call a confe

Re: [asterisk-users] how to send dtmf after pause ?

n play the dtmf tones and bridge the call to your extension afterwards. yves Am 07.06.2013 17:51, schrieb Sean Darcy: I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#

[asterisk-users] db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database ??

I'm showing a lot of these on the console. I'm not using any database. Where would this be coming from? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introd

[asterisk-users] no silk translation ?

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551...@voice.google.com-da3c [Jun 10 16:18:22] WARNING[40

Re: [asterisk-users] no silk translation ?

On 06/10/2013 05:24 PM, Sean Darcy wrote: Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551

Re: [asterisk-users] no silk translation ?

On 06/10/2013 05:24 PM, Sean Darcy wrote: Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551

[asterisk-users] 11.4.0: iax packets lost by amazon ec2

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

On 09/06/2013 07:08 PM, Steve Edwards wrote: On Fri, 6 Sep 2013, Sean Darcy wrote: I'm not sure asterisk is even listening for the packets: [root@asterisk ~]# netstat -apnt | grep 4569 [root@asterisk ~]# '-t' meand TCP. IAX is UDP. My bad: netstat -apnu | grep 4569 udp

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

On 09/07/2013 10:33 AM, Tony Mountifield wrote: In article <522a934d.8010...@gmail.com>, Sean Darcy wrote: On 09/06/2013 07:08 PM, Steve Edwards wrote: On Fri, 6 Sep 2013, Sean Darcy wrote: I'm not sure asterisk is even listening for the packets: [root@asterisk ~]# netstat -

Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

On 09/07/2013 01:26 PM, Tony Mountifield wrote: In article , Sean Darcy wrote: On 09/07/2013 10:33 AM, Tony Mountifield wrote: In article <522a934d.8010...@gmail.com>, Sean Darcy wrote: On 09/06/2013 07:08 PM, Steve Edwards wrote: On Fri, 6 Sep 2013, Sean Darcy wrote: I'

[asterisk-users] permission problems on amazon ec2

I'm marching forward trying to get asterisk running on a amazon EC2 instance, Fedora 19. If I start it from the terminal all works. I can login as user "asterisk" and start asterisk. But if I try to use systemctl to start it automatically I get the error it doesn't have the permission to op

[asterisk-users] iax2: two users can't authenticate from same ip address

I'm trying set up asterisk on an amazon instance in Sydney. It's to use for our kids in Sydney to connect with their friends in the States. We've found iax works better than sip with these distances. But we now have weird problem: everybody has a cell phone, and it's much cheaper/better to use

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

On 09/09/2013 08:04 AM, Julian Beach wrote: Hello Sean, Sunday, September 8, 2013, 11:25:24 PM, you wrote: The problem is that once a phone has used the server, no other phone can use it. The servers sees all the phones as having the same ip address (though different ports). This sounds li

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

On 09/09/2013 11:08 AM, Joshua Colp wrote: Sean Darcy wrote: On the server each device has type=friend. I do notice that peer "home" has the standard iax port 4569. The other peers are assigned 1026, 1027 and 1028. How are these ports assigned? The actual configuration entr

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

On 09/09/2013 01:54 PM, Joshua Colp wrote: Sean Darcy wrote: home is from the home machine, which registers with the server: register => home:@ [home] type=friend insecure=port,invite secret= ; same secret as on server context=incoming host= You aren't specifying what use

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy Sent: Monday, September 09, 2013 3:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip address Dial("IAX2/home-143

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

ESTABLISHED -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy Sent: Monday, September 09, 2013 7:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] iax2: two users can't authentic

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

On 09/10/2013 05:27 PM, Joshua Colp wrote: Sean Darcy wrote: On 09/10/2013 12:15 PM, Joshua Colp wrote: Sean Darcy wrote: Maybe a different question would be helpful. Let's assume no NAT; the server is directly connected with an FQDN. Two iax devices register. Does asterisk assign

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

On 09/10/2013 12:15 PM, Joshua Colp wrote: Sean Darcy wrote: Maybe a different question would be helpful. Let's assume no NAT; the server is directly connected with an FQDN. Two iax devices register. Does asterisk assign them different ports? Asterisk does not assign ports. The IAX2 ch

[asterisk-users] iax: unable to transfer - one way audio

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from : > requested format = speex, > requested prefs = (), > actual form

Re: [asterisk-users] iax: unable to transfer - one way audio

On 09/27/2013 09:08 PM, Sean Darcy wrote: We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from : > requested format

Re: [asterisk-users] iax: unable to transfer - one way audio

On 09/28/2013 11:11 AM, Asghar Mohammad wrote: Hi, If you post your configuration someone may help you. On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy mailto:seandar...@gmail.com>> wrote: On 09/27/2013 09:08 PM, Sean Darcy wrote: We have zoiper connected over iax to aster

Re: [asterisk-users] iax: unable to transfer - one way audio

On 09/30/2013 12:09 PM, Sean Darcy wrote: On 09/28/2013 11:11 AM, Asghar Mohammad wrote: Hi, If you post your configuration someone may help you. On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy mailto:seandar...@gmail.com>> wrote: On 09/27/2013 09:08 PM, Sean Darcy wrote: W

[asterisk-users] iax2: no authentication, but still peer?

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed authentication. The secret seems correct, so we can't figure out why we're getting failed authentication. But at the same time the device shows as registered: [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_he

Re: [asterisk-users] iax2: no authentication, but still peer?

On 10/08/2013 03:29 PM, Adrian Serafini wrote: The qualify is on for the peer. It is failing to reply to the requested SIP status. Maybe it is on wifi, screen goes off, wifi follows, zoiper iax stack doesn't re-reg with the asterisk. [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_proc

[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] 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

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: 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: 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] 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] 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] 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

[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] 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

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

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

[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 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 -- __

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

[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

[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

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

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 01/02/2012 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] 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] 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] 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] best softphone for 2012?

On Sat, Jan 7, 2012 at 9:34 AM, Gilles wrote: > On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy > wrote: >>But what really made us choose linphone was you use it on android/iphone. >> >>That has been a huge plus. As a bonus, you can use any degegistered >>smartphon

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] 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

[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

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

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

[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] 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] 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

[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

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] 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] 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

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 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

[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 -- __

[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

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] 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

[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] Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser

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

sean darcy wrote: > I'm trying to setup sipgate on 1.6.1. Following the instructions on the > site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, > > I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: > > [sipgate]

Re: [asterisk-users] Libpri-1.4.10.2 Released

Karl Fife wrote: > Question about the proper way to update LibPRI: > > 'Bouncing' asterisk after an installing the new LibPRI version does > indeed reflect the update: > asterisk*CLI> pri show version > libpri version: 1.4.10.2 > Hmm. What asterisk version are you running? On 1.6.0.18-rc2: pbx

[asterisk-users] 1.6.0.18-rc3: SendFAX causes restart

On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX asterisk restarts: [Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing [...@fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m("ESC[1;35;40mSIP/nhi-rive rside-sip-ESC[0;37;40m", "ESC[1;35;40mContext fax-tx-testESC[0

Re: [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart

Leif Madsen wrote: > sean darcy wrote: >> On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX >> asterisk restarts: >> >> Before I file a bug, is there anything I'm missing? > > Does this happen on earlier versions of the 1.6.0 series prio

Re: [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart

sean darcy wrote: > Leif Madsen wrote: >> sean darcy wrote: >>> On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX >>> asterisk restarts: >>> >>> Before I file a bug, is there anything I'm missing? >> Does this happ

[asterisk-users] Setting up Nokia e71: registration problem

In SIP setting on the e71 I set the public user name as 1...@10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[1

[asterisk-users] transferring SIP call: no voice

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax.

Re: [asterisk-users] transferring SIP call: no voice

sean darcy wrote: > I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk > B. Both are behind NAT, but port forwarded. I get the connection, but no > voice - either in or out. > > I can call on SIP from A to B (and from B to A). Do it all the time. >

[asterisk-users] DAHDI/1-2 v. DAHDI/2-1 ??

I've got a single TDM 400P board with two internal ports and 1 external. chan_dahdi.conf: context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS group=0 channel => 1 ; Telephone attached to port 1 channel

Re: [asterisk-users] DAHDI/1-2 v. DAHDI/2-1 ??

sean darcy wrote: > I've got a single TDM 400P board with two internal ports and 1 external. > > chan_dahdi.conf: > > context=internal ; Uses the [internal] context in extensions.conf > signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS &

Re: [asterisk-users] Asterisk to Email

On Sun, Dec 6, 2009 at 12:28 PM, Thomas Perron wrote: > I am reading a lot of the material but need your input to help me > understand what you mean. > > System(echo body of message | mail -s "subject line" > ${the_caller_...@tmobile.net) > > I understand the System application generally > echo bo

Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)

listu...@spamomania.co.uk wrote: > On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: >> Appears completely resolved! >> No more home-spun patches! >> Thanks! >> -K >> > It's *not* fixed here: > DAHDI Version: 2.2.1 Echo Canceller: MG2 > > But as is depressingly the 'norm' for Asterisk it comes b

[asterisk-users] callerid not working over sip

Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [...@internal:2] NoOp("DAHDI/1-1", "Co

Re: [asterisk-users] callerid not working over sip

sean darcy wrote: > Calling from my home using Asterisk 1.6.2.1 to an office extension > (Asterisk 1.6.1.13) the callerid is not honored: > > Home: > > -- Starting simple switch on 'DAHDI/1-1' > -- Executing [...@internal:1] Answer("DAHDI/1-1&qu

Re: [asterisk-users] callerid not working over sip

Steve Howes wrote: > On 31 Jan 2010, at 16:24, sean darcy wrote: >>> -- Executing [...@internal:3] Set("DAHDI/1-1", "CALLERID="Test" >>> <447>") in new stack >>> >>> Why isn't the office asterisk picking up the

<    1   2   3   4   5   6   >