[asterisk-users] how do I run a command on "Failed to authenticate" ?

16.13.0, pjsip I'd like to get an alert if a call fails to authenticate: if "Failed to authenticate" then mail someone the source ip endif As I look at ami or ari, they deal with calls in channels. Is there a way to get failed invites or registers ? sean --

Re: [asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

On 9/7/20 3:41 PM, Joshua C. Colp wrote: On Sat, Sep 5, 2020 at 10:23 AM sean darcy <mailto:seandar...@gmail.com>> wrote: module load res_pjsip Unable to load module res_pjsip Command 'module load res_pjsip' failed. ERROR[141535]: loader.c:281 module_load_e

Re: [asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

On 9/6/20 3:43 AM, Michael Maier wrote: On 05.09.20 at 15:22 sean darcy wrote: asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: This means, that your pjsip library doesn't match the asterisk binary. It's best to remove the independent pjsip libr

[asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4

Re: [asterisk-users] permission woes on systemd

On 1/22/20 11:51 AM, Michael L. Young wrote: - Original Message - From: "sean darcy" To: "Asterisk Users Mailing List, Non-Commercial Discussion" Sent: Tuesday, January 21, 2020 9:22:28 PM Subject: [asterisk-users] permission woes on systemd [..] So why wou

[asterisk-users] permission woes on systemd

I'm running asterisk 16 on Fedora 31. If I start asterisk as user asterisk, all goes well. But if I start asterisk from systemd: asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 sorcery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk

Re: [asterisk-users] USB dahdi fxo ?

On 12/14/19 11:29 AM, Greg Troxel wrote: sean darcy writes: There is also the ObiHai OBi202 with an OBiLine, which provides an FXO port remoted over SIP. (I am not sure if this is discontinued.) "FXO port remoted over SIP"? I have an analog phone system. I can use the obi202

Re: [asterisk-users] USB dahdi fxo ?

On 12/13/19 9:28 PM, Greg Troxel wrote: sean darcy writes: I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? There is also the Obi

[asterisk-users] USB dahdi fxo ?

I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? sean -- _ -- Bandwidth and

Re: [asterisk-users] trouble building dahdi on kernel 5.2.7

On 8/14/19 6:00 PM, sean darcy wrote: dahdi built fine on 5.1.20, but on 5.2.7: .   CC [M] /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o   SHIPPED /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader

[asterisk-users] trouble building dahdi on kernel 5.2.7

dahdi built fine on 5.1.20, but on 5.2.7: . CC [M] /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o SHIPPED /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o LD [M] /home/asterisk/r

Re: [asterisk-users] pjsip endoint woes

On 4/9/19 12:14 PM, George Joseph wrote: On Tue, Apr 9, 2019 at 9:28 AM sean darcy <mailto:seandar...@gmail.com>> wrote: On 4/8/19 6:18 AM, Joshua C. Colp wrote: > On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: >> On 4/5/19 10:36 AM, sean darcy wrote:

Re: [asterisk-users] pjsip endoint woes

On 4/8/19 6:18 AM, Joshua C. Colp wrote: On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: On 4/5/19 10:36 AM, sean darcy wrote: I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type

Re: [asterisk-users] pjsip endoint woes

On 4/5/19 10:36 AM, sean darcy wrote: I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = [obi202-aor](!) type = aor max_contacts = 2 ; = endpoints  ===

[asterisk-users] pjsip endoint woes

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = [obi202-aor](!) type = aor max_contacts = 2 ; = endpoints [gv-voice](obi202-endpoint) auth = gv-voice aor

Re: [asterisk-users] why doesn't extension "s" work ?

s", those calls will match whatever the dialed number is. On 03/28/2019 08:32 PM, sean darcy wrote: I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}

[asterisk-users] why doesn't extension "s" work ?

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n, But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_

[asterisk-users] pattern matching "+"

From my provider I get extensions of: +1<10digit number> 1<10 digit number> <10 digit number> seemingly randomly. What I'd like to do is exten=_!1234567890,1,Answer() which would match anything ending in 1234567890. But that doesn't work since ! can only be used at the end of a pattern. I t

Re: [asterisk-users] trouble removing + sign

On 2/14/19 4:23 AM, Administrator TOOTAI wrote: Le 14/02/2019 à 00:12, sean darcy a écrit : I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried:   ;  strip leading plus sign    same => n,

Re: [asterisk-users] trouble removing + sign

On 2/13/19 6:22 PM, Dovid Bender wrote: Try == in your gotoif (instead of =) Regards, Dovid   Original Message From: seandar...@gmail.com Sent: February 14, 2019 01:14 To: asterisk-users@lists.digium.com Reply-to: asterisk-users@lists.digium.com Subject: [asterisk-users] troubl

[asterisk-users] trouble removing + sign

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLE

[asterisk-users] what service does asterisk need to avoid netsock error ?

I'm running Fedora 29. asterisk starts with a systemd service at boot. On any reboot I get a LOT of : [Jan 15 09:30:26] ERROR[1162]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. [Jan 15 09:30:35] ERROR[1161]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. [Jan 1

[asterisk-users] Surprise: DAHDI 3.0.0. Analog TDM cards EOL ?

I must have missed the memo, but I was surprised to see a new DAHDI release in downloads. Was there an announcement ? Is there a Changelog ? Also, it seems there's no longer a wctdm module. What's the plan for the analog TDM cards ? Or is there one ? sean --

[asterisk-users] continuous netsock errors

I get continuous errors about "unknown address family" : ERROR[1233]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. ERROR[1233]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. ERROR[1226]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. ...

Re: [asterisk-users] Is there any way to pass caller id to

On 10/16/18 1:42 PM, Antony Stone wrote: On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote: Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it sho

Re: [asterisk-users] Convert SIP to PJSIP

On 9/24/18 2:57 PM, John T. Bittner wrote: Hello all, I am having some trouble converting this setup from SIP to PJSIP. Any help is much appreciated. I used the converter script and get most of it but don’t see a registration entry. How do you convert this entry into PJSIP. This working s

Re: [asterisk-users] Convert SIP to PJSIP

On 9/24/18 5:04 PM, John T. Bittner wrote: Hello all, I am having some trouble getting this to work under pjsip. Any help is much appreciated. I used the converter script and I see it register but can’t receive or send to ringcentral. Anyone get this working with PJSIP? Works with chan_si

Re: [asterisk-users] hangup the _called_ channel ?

On 9/12/18 1:32 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote: On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x

Re: [asterisk-users] hangup the _called_ channel ?

On 9/12/18 1:25 PM, sean darcy wrote: On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP

Re: [asterisk-users] hangup the _called_ channel ?

On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file

Re: [asterisk-users] hangup the _called_ channel ?

On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file

[asterisk-users] hangup the _called_ channel ?

I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, t

[asterisk-users] STUN re-evalutation every 2 minutes ??

13.21.0 Every 2-3 minutes: Sep 1 16:00:57] WARNING[150257]: res_stun_monitor.c:140 stun_monitor_request: STUN poll got no response. Re-evaluating STUN server address. [Sep 1 16:02:18] NOTICE[150257]: res_stun_monitor.c:151 stun_monitor_request: Old external address/port :42562 now seen as

[asterisk-users] Community forum ?

I see a lot of tag lines on posts for the Asterisk Community Forum. Is that forum supposed to supersede this mailing list ? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up

Re: [asterisk-users] getting invites to rtp ports ??

when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: On 08/29/2018 08:07 PM, John Covici wrote: I wonder if I could have that patch, maybe I could add it to my fail2ban regexp and if you have the correct regexp, I would apperciate that as well. Thanks. On

Re: [asterisk-users] getting invites to rtp ports ??

better than nothing. Digium warns not to use fail2ban / log trolling as a security system: http://forums.asterisk.org/viewtopic.php?p=159984 -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Wednesday, August 29, 2018 6

Re: [asterisk-users] getting invites to rtp ports ??

- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Wednesday, August 29, 2018 10:46 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] getting invites to rtp ports ?? On 08/29/2018 09:42 AM, Carlos Rojas wrote: Hi Probably

Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 09:42 AM, Carlos Rojas wrote: Hi Probably somebody is trying to hack your system, you should block that ip on your firewall. Regards On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <mailto:seandar...@gmail.com>> wrote: I'm getting invites to very high ports ev

[asterisk-users] getting invites to rtp ports ??

I'm getting invites to very high ports every 30 seconds from a particular ip address: Retransmitting #10 (NAT) to 5.199.133.128:52734: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 From: ;tag=1872048972 To: ;tag=as3a52e748 C

Re: [asterisk-users] Decoding SIP register hack

On 05/17/2018 05:29 PM, sean darcy wrote: On 05/17/2018 04:47 PM, Daniel Tryba wrote: On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:     WARNING.* .*: fail2ban='' # Option:  ignoreregex # Notes.:  regex to ignore. If this regex matches, the line is ignored. # Val

Re: [asterisk-users] Decoding SIP register hack

On 05/17/2018 04:47 PM, Daniel Tryba wrote: On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote: WARNING.* .*: fail2ban='' # Option:  ignoreregex # Notes.:  regex to ignore. If this regex matches, the line is ignored. # Values:  TEXT # ignoreregex = Th

Re: [asterisk-users] Decoding SIP register hack

On 05/17/2018 11:38 AM, Frank Vanoni wrote: On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: 3. How do I set up the server to block these ? 4. Can I stop the retransmitting of the 401 Unauthorized packets ? I'm happy with Fail2Ban protecting my Asterisk 13. Here is my configuration

[asterisk-users] Decoding SIP register hack

I need some help understanding SIP dialog. Some actor is trying to access my server, but I can't figure out what he's trying to do ,or how. I'm getting a lot of these warnings. [May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission _zIr9tD

Re: [asterisk-users] opus from git : install questions

On 02/06/2018 07:58 AM, Tzafrir Cohen wrote: On Sun, Feb 04, 2018 at 03:15:02PM -0500, sean darcy wrote: On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you

Re: [asterisk-users] OAuth : xmpp.conf

On 02/03/2018 07:11 PM, sean darcy wrote: Confused about xmpp.conf with OAuth. Let's assume I have two voice accounts. Are all the OAuth entries in each account ? It'd be really great if only separate refresh_token s were required! For instance- painful: [general] ...

[asterisk-users] opus from git : install questions

On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you do not have to (re-) make your whole Asterisk. The Makefile generates: codecs/codec_opus_open_source.so f

[asterisk-users] OAuth : xmpp.conf

Confused about xmpp.conf with OAuth. Let's assume I have two voice accounts. Are all the OAuth entries in each account ? It'd be really great if only separate refresh_token s were required! For instance- painful: [general] ... [gv1] . refresh_token=gv1-token oauth_clientid=gv1-client-

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 12/30/2017 08:18 PM, Dovid Bender wrote: Script kiddies trying to find vulnerable systems that they can make calls on. Lock down the box with iptables and use fail2ban to block them. The via is probably bogus unless a box at the DoD was comprimised. On Sat, Dec 30, 2017 at 6:49 PM, sean

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 12/30/2017 08:10 PM, Antony Stone wrote: On Sunday 31 December 2017 at 00:49:17, sean darcy wrote: I've been getting a lot of timeouts on non-critical invite transactions. So how is someone on a Dutch ISP using my server to mess with a US DoD ip address ? What's your s

[asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4

Re: [asterisk-users] How to set outgoing sip callid ?

On 05/31/2016 11:43 AM, Frank Vanoni wrote: (CALLERID(all)="Jon Doe" <+123456789>) So simple. just too obvious. thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

[asterisk-users] How to set outgoing sip callid ?

Calling linphone from asterisk 13.9.1.: Dial(SIP/@sip.linphone.org) And it works. But on the linphone side the caller is: @ipaddress or 2502@45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPC

[asterisk-users] "__sip_xmit....Success" every 15 seconds !

On 13.9. The cli log has these messages every 15 seconds. The end point to linphone on android. [May 12 19:02:59] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7effe40088b0 (len 608) to 10.10.11.95:37855 returned -2: Success [May 12 19:03:13] WARNING[2555]: chan_sip.c:37

[asterisk-users] codec_opus w/ PLR and FEC for Asterisk 11

There is an opus patch for asterisk 11. https://github.com/seanbright/asterisk-opus/tree/asterisk-11 . But it doesn't have Packet Loss Resilience or Forward Error Correction, both of which are important for voip. 2.1.6. Packet Loss Resilience Audio codecs often exploit inter-frame correlati

Re: [asterisk-users] opus : patches for FEC and PLC useful ?

On 04/05/2016 04:17 AM, Tzafrir Cohen wrote: On Mon, Apr 04, 2016 at 11:39:07AM +0200, Ludovic Gasc wrote: We're testing this branch for a while, not with the latest commits. For now, it works, however, time to time audio quality issues with transcoding, but I don't know yet where is the proble

[asterisk-users] opus : patches for FEC and PLC useful ?

In a fork of seanbright's opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip: https://github.com/traud/asterisk-opus Anybody used these patches ? Puzzled why they weren't committed to the main patch

Re: [asterisk-users] registering IAX with Teliax

On 05/13/2015 03:51 PM, Greg Woods wrote: Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from

Re: [asterisk-users] PJProject Bundled Update

On 03/31/2016 11:57 AM, George Joseph wrote: As you know, the ability to use a bundled version of pjproject was introduced with Asterisk 13.8.0. More info on the Asterisk Wiki a

Re: [asterisk-users] asterisk a "less secure app" on google ??

;t tried it with this exact setup though. On Sun, Mar 27, 2016 at 6:13 AM, sean darcy mailto:seandar...@gmail.com>> wrote: To connect to google voice with xmpp, I've had to turn on the "less secure apps" switch. You recently changed your security settings so th

[asterisk-users] asterisk a "less secure app" on google ??

To connect to google voice with xmpp, I've had to turn on the "less secure apps" switch. You recently changed your security settings so that your Google Account ...@gmail.com is no longer protected by modern security standards. Please be aware that it is now easier for an attacker to brea

Re: [asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?

end calls from my in-state area codes to the 'RING' label, other calls are answered by voicemail. On Thu, Feb 25, 2016 at 3:13 PM, sean darcy mailto:seandar...@gmail.com>> wrote: I'd like to transfer all my pesky telemarketing calls to Jolly Roger . http://www.nyt

[asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?

I'd like to transfer all my pesky telemarketing calls to Jolly Roger . http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html In the middle of a call I'd hit some DTMF sequence, which would dial Jolly Roger and transfer the call after Jolly Roger answers.

Re: [asterisk-users] 11.21.0 : echo woes : can't install canceller (sean darcy)

On 01/29/2016 03:59 PM, Mc GRATH Ricardo wrote: Hi Sean Darcy Question about "the remote party always hears an echo on it's side", strange because eco suppression circuit is for local side. Mc GRATH Ricardo OK. Maybe an echo canceller won't make any difference. But

Re: [asterisk-users] 11.21.0 : echo woes : can't install canceller

On 01/28/2016 03:39 PM, sean darcy wrote: i've got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions. Using an analogue internal phone, the remote party always hears an echo on it's side. We do not hear an echo. Doesn't ma

[asterisk-users] 11.21.0 : echo woes : can't install canceller

i've got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions. Using an analogue internal phone, the remote party always hears an echo on it's side. We do not hear an echo. Doesn't matter who is the calling party. But if we use a SIP exten

[asterisk-users] map softphone ids to asterisk ??

In setting up the GS-Wave softphone there are two id entries: SIP User ID SIP Authentication ID I would have thought SIP User ID was the devicename , i.e. [name]. Then SIP Authentication ID was defaultuser. But not so. With [gs_5062](cell-phones) defaultuser=gs_62 and SIP User ID

Re: [asterisk-users] no ringing tone with Dial option r

On 11/03/2015 01:11 PM, John Kiniston wrote: Have you checked your indications.conf? I've seen a missing or misconfiguration in the zone definition cause this. On Tue, Nov 3, 2015 at 11:07 AM, sean darcy mailto:seandar...@gmail.com>> wrote: On 11/01/2015 12:38 PM, sean

Re: [asterisk-users] no ringing tone with Dial option r

On 11/04/2015 03:40 AM, A J Stiles wrote: On Tuesday 03 Nov 2015, sean darcy wrote: On 11/01/2015 12:38 PM, sean darcy wrote: I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Otherwise

Re: [asterisk-users] no ringing tone with Dial option r

On 11/04/2015 03:43 AM, Bertrand LUPART - Linkeo.com wrote: Hello, I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Warning, options are the 3rd arguments. You seem to have an extra comma and a non-closed doub

Re: [asterisk-users] no ringing tone with Dial option r

On 11/01/2015 12:38 PM, sean darcy wrote: I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean FWIW, 11

[asterisk-users] no ringing tone with Dial option r

I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean -- _ -- Bandwidth and Colocation

[asterisk-users] more vm woes

on 11.17.1 Trying to debug vm. as suggested by voip-info: mailcmd=cat >> /tmp/astvm-mail and the mailbox is: 1143 => ,sean,s...@company.com,,|tz=eastern|attach=yes|saycid=yes But there [s nothing in /tmp. The messages are in /var/spool/asterisk/voicemail/43-vm/1143/ . Any help appreci

Re: [asterisk-users] howto copy a voicemail message to another machine ?

On 06/16/2015 11:52 AM, D'Arcy J.M. Cain wrote: On Tue, 16 Jun 2015 11:35:26 -0400 sean darcy wrote: My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. Really? That

[asterisk-users] howto copy a voicemail message to another machine ?

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message.

Re: [asterisk-users] Free Fax for Asterisk : registered, but license not effective

On 06/05/2015 08:15 PM, sean darcy wrote: I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXX.lic

[asterisk-users] Free Fax for Asterisk : registered, but license not effective

I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXX.lic but: Executing [17772822954@FaxIncoming:4] Rece

[asterisk-users] 11.17.1 : ReceiveFax then signal 11 ??

dialplan [FaxIncoming] exten=s,1,NoOp(Incoming fax on 46-va) same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)}) same=n,Answer() same=n,ReceiveFAX(${FAXFILE}.tif,df) same=n,Hangup() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERRO

[asterisk-users] did i miss the memo on asterisk devel ?

I usually lurk on the asterisk devel list to see what's going on. No posts for a week or two. Has the list moved ? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] dahdi 2.10.1 build fails

On 06/01/2015 04:49 PM, Shaun Ruffell wrote: On Mon, Jun 01, 2015 at 04:24:31PM -0400, sean darcy wrote: On fedora 21, trying to build dahdi for kernel 4.0.4. gcc-4.9.2-6.fc21.x86_64 make -C /lib/modules/4.0.4-202.fc21.x86_64/build SUBDIRS=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1

[asterisk-users] dahdi 2.10.1 build fails

On fedora 21, trying to build dahdi for kernel 4.0.4. gcc-4.9.2-6.fc21.x86_64 make -C /lib/modules/4.0.4-202.fc21.x86_64/build SUBDIRS=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi DAHDI_INCLUDE=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/include DAHDI_MODULES_EXTRA=" "

[asterisk-users] "timeout on non-critical invite" spamming log

On 11.17.1: The cli and the log are full of these warnings: WARNING[12110]: chan_sip.c:4086 retrans_pkt: Timeout on 849421411 on non-critical invite transaction. The number is a random 9-10 digits. What causes them? How do I stop them ? sean -- ___

[asterisk-users] which libsrtp ?

I've been having some issues with srtp. so I checked which version of libsrtp I built asterisk 11.6 against. I'm on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64. From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, released a couple of weeks ago. I'm not a f

[asterisk-users] Upgrade to Fedora 21, now gv requires rtp ?

I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works with Fedora 20. -- Executing [s@DialOut:15] Dial("DAHDI/1-1", "motif/8447/+1212xxxy...@voice.google.com,,rTt") in new stack [Mar 1 21:24:06] ERROR[2477][C-]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was

Re: [asterisk-users] 11.5.0: blindxfer problems

On 12/22/2014 05:38 PM, Richard Mudgett wrote: On Mon, Dec 22, 2014 at 4:00 PM, sean darcy mailto:seandar...@gmail.com>> wrote: How do I enable DTMF logging? logger set level DEBUG No such command 'logger set level DEBUG' (type 'core show help logger

Re: [asterisk-users] Smartphone Mobility App?

On 12/19/2014 09:29 AM, chris wrote: Anyone found any good smartphone apps that connect with their asterisk boxes that provides basic mobility features? The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their

Re: [asterisk-users] 11.5.0: blindxfer problems

On 12/21/2014 11:09 AM, sean darcy wrote: On 12/21/2014 04:42 AM, Patrick Beaumont wrote: Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using “sip info” for the DTMF signalling as the RFC signalling was

Re: [asterisk-users] 11.5.0: blindxfer problems

cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" wrote: On 12/20/2014 03:22 PM, sean darcy wrote: On 12/19/2014 09:42 AM, Rusty Newton wrote: On Wed, Dec 17, 2014 at 1:09 PM, sean darcy wrote: I've got a confbridge set up

Re: [asterisk-users] 11.5.0: blindxfer problems

On 12/20/2014 03:22 PM, sean darcy wrote: On 12/19/2014 09:42 AM, Rusty Newton wrote: On Wed, Dec 17, 2014 at 1:09 PM, sean darcy wrote: I've got a confbridge set up which works if dialed locally: -- Executing [266@internal:1] Answer("DAHDI/1-1", "") in new sta

Re: [asterisk-users] 11.5.0: blindxfer problems

On 12/19/2014 09:42 AM, Rusty Newton wrote: On Wed, Dec 17, 2014 at 1:09 PM, sean darcy wrote: I've got a confbridge set up which works if dialed locally: -- Executing [266@internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266@internal:2] Send

[asterisk-users] 11.5.0: blindxfer problems

I've got a confbridge set up which works if dialed locally: -- Executing [266@internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266@internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266@internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- Playing '

Re: [asterisk-users] On Fedora, kernel update resets /var/run/asterisk owner to root.root

On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: On Tue, Dec 2, 2014 at 1:22 PM, sean darcy wrote: Or do I find a new place to put asterisk.pid? Also, if you use the native systemd unit file, you no longer need a PID file, although you still need /run/asterisk to store the control socket. So

[asterisk-users] On Fedora, kernel update resets /var/run/asterisk owner to root.root

On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean -- ___

Re: [asterisk-users] Asterisk 13 : SILK codec ?

On 10/29/2014 08:06 PM, Matthew Jordan wrote: On Wed, Oct 29, 2014 at 5:16 PM, sean darcy wrote: Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A number of performance improvements in the media

[asterisk-users] Asterisk 13 : SILK codec ?

Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )

On 10/24/2014 03:24 PM, Jeffrey Ollie wrote: On Fri, Oct 24, 2014 at 1:47 PM, sean darcy wrote: On 10/24/2014 02:21 PM, Jeffrey Ollie wrote: restorecon -rv /etc/asterisk I'd never have guessed. Yeah, if you "mv" the data instead of "cp" the data from one place

Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )

On 10/24/2014 02:21 PM, Jeffrey Ollie wrote: restorecon -rv /etc/asterisk I'd never have guessed. Thanks. I owe you a beer. At least one. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )

On 10/23/2014 01:19 PM, sean darcy wrote: On 10/23/2014 11:26 AM, sean darcy wrote: Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP chann

Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )

On 10/23/2014 11:26 AM, sean darcy wrote: Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]:

[asterisk-users] 11.13.1: unable to load sip.conf (or iax )

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config s

Re: [asterisk-users] On kernel 3.16.2 : dahdi_rec: Invalid argument

On 09/13/2014 01:52 PM, sean darcy wrote: On 09/13/2014 12:09 PM, sean darcy wrote: On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against it. dahdi show channels works fine, but when I try to place a call: chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument Any help

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