[asterisk-users] VoIP support engineer opportunity

A full CV is welcome. Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] RTP address learning and timing problem

happening. > > On Mon, Apr 17, 2023 at 8:52 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi Joshua, >> >> Thank you for that. From the code it kind of looks like >> STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:

Re: [asterisk-users] RTP address learning and timing problem

read the logic[1]. There's an entire comment > that talks about how it works. > > [1] > https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 > > On Mon, Apr 17, 2023 at 7:10 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> H

Re: [asterisk-users] RTP address learning and timing problem

you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote: > On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote: > >> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >> dcunning...@voisonics.com> wrote: >> >>> Hello, >>> >>> Does a

Re: [asterisk-users] RTP address learning and timing problem

Thank you Joshua. On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote: > On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote: > >> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >> dcunning...@voisonics.com> wrote: >> >>> Hello, >>> >&g

Re: [asterisk-users] RTP address learning and timing problem

Hi Joshua, Thanks for that. The naming is a little confusing as "no'' makes it sound like it's "not strict" - good to know though. Is it possible to set strictrtp to no for just one peer? On Wed, 1 Mar 2023 at 02:57, Joshua C. Colp wrote: > On Tue, Feb 28, 2023 at 9:50 A

Re: [asterisk-users] RTP address learning and timing problem

Hello, Does anyone know if one of the "strictrtp" options disables RTP learning? As far as I can tell from the documentation the values "no" and "seqno" are more permissive in allowing other sources rather than less, but I thought I'd check. Thanks. On Th

[asterisk-users] RTP address learning and timing problem

o the problem. Can anyone suggest how to prevent this problem? Is it possible to turn off learning the media address per call or per peer? Thanks for your help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand:

Re: [asterisk-users] mailing list working?

Yes, the technical part of the list is working, but I'm not as sure about the functional part. (Meaning that several people's questions are going unanswered.  I wish that I had the answers they are looking for, but alas, I don't.) Dave On 1/16/2023 6:08 AM, marek wrote: there are new

Re: [asterisk-users] Voicemail Transcription with openai/whisper

I really love this idea. Thanks for sharing. I've been looking for a good way to provide this service to my customers. Hopefully this will work for me too. Thanks, Dave On 11/27/2022 8:08 AM, Doug Lytle wrote: Everybody, I've recently discovered openai/whisper and have been trying in

Re: [asterisk-users] CSeq reset on re-INVITE

Okay, thanks very much for your help Joshua. On Mon, 31 Oct 2022 at 10:07, Joshua C. Colp wrote: > On Sun, Oct 30, 2022 at 5:00 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi Joshua, >> >> Thanks very much. I presume this is the relevant

Re: [asterisk-users] CSeq reset on re-INVITE

ckets from the Asterisk log. Can anyone see an issue that would cause the error? Thanks in advance. On Sat, 29 Oct 2022 at 12:03, Joshua C. Colp wrote: > On Fri, Oct 28, 2022 at 6:28 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hello, >> >> We

[asterisk-users] CSeq reset on re-INVITE

error is that call 1 was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk resetting the CSeq on the re-INVITE, and doesn't this appear to be incorrect? Thanks in advance for any help. --

[asterisk-users] VoIP support engineer opportunity

A full CV is welcome. Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] Asterisk "we couldn't allocate a port for RTP" errors

t; asterisk a restart... :S > > On Wed, Jul 27, 2022 at 6:21 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hello, >> >> We have an Asterisk 13.38.2 server which today started giving "we >> couldn't allocate a port for RTP" errors.

[asterisk-users] Asterisk "we couldn't allocate a port for RTP" errors

. Thanks in advance for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] TCP dial via proxy

using ? > please show: asterisk -rx "sip show peer sip-peer" > > I checked... > I use UDP and TCP, my phone via UDP, telekom via TCP and works > > > same => n,dial(SIP/${EXTEN}@sip-trunk-telekom) > > [image: image.png] > > > On Thu, 21 Jul 2022 at 23:

Re: [asterisk-users] TCP dial via proxy

k-users@lists.digium.com> > *Date: *Thursday, July 21, 2022 at 9:21 AM > *To: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Subject: *Re: [asterisk-users] TCP dial via proxy > > > > David, > > > > We had this e

Re: [asterisk-users] TCP dial via proxy

Hi Dovid, Thanks for the reply. We are indeed able to force TCP from the Kamailio proxy, but haven't been able to force it between Asterisk and Kamailio. On Fri, 22 Jul 2022 at 01:21, Dovid Bender wrote: > David, > > We had this exact "issue" in the past and were not abl

Re: [asterisk-users] TCP dial via proxy

= final.destination.com transport = tcp outboundproxy = our.proxy.com On Fri, 22 Jul 2022 at 01:23, Henning Follmann wrote: > On Thu, Jul 21, 2022 at 02:46:07PM +1200, David Cunningham wrote: > > Hello, > > > > We have an Asterisk dial which sends the call via a proxy usin

[asterisk-users] TCP dial via proxy

hat didn't seem to work. We are using chan_sip. Thanks very much for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocatio

Re: [asterisk-users] Listen on 2 of 3 IP addresses

Thank you Thomas. On Mon, 18 Jul 2022 at 12:24, Thomas Ray wrote: > Moving to chan_pjsip solves this problem. > > > > *From: *asterisk-users on > behalf of David Cunningham > *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-user

Re: [asterisk-users] Listen on 2 of 3 IP addresses

ote: > > On Fri, Jul 15, 2022 at 1:37 AM David Cunningham < > dcunning...@voisonics.com> > > wrote: > > > > > Hello, > > > > > > We have an Asterisk server with 3 IP addresses, and need to listen on > only > > >

[asterisk-users] Listen on 2 of 3 IP addresses

much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] No audio for 10 seconds and then comfort noise

Hi Joshua, You're right, it was a firewall problem. One of those things where testing a change in one place throws up a previously unseen problem somewhere else! Thanks for the tip. On Thu, 19 May 2022 at 21:18, Joshua C. Colp wrote: > On Thu, May 19, 2022 at 6:04 AM David Cunning

Re: [asterisk-users] No audio for 10 seconds and then comfort noise

022 at 3:52 AM Dovid Bender wrote: > >> David, >> >> Are you getting any RTP from the PSTN for either leg? If not it could be >> that they assume you are behind NAT and want to see where the SRC of the >> RTP before they send it back. We had a few carriers that

Re: [asterisk-users] No audio for 10 seconds and then comfort noise

DI running on the server: # asterisk -rx 'dahdi show version' DAHDI Version: 3.0.0 Echo Canceller: # asterisk -rx 'dahdi show status' Description Alarms IRQbpviol CRCFra Codi Options LBO On Thu, 19 May 2022 at 15:51, David Cunningham wrote: > Hello, >

[asterisk-users] No audio for 10 seconds and then comfort noise

nt RTP packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160) [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160) Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com

Re: [asterisk-users] Context for 302 Moved response

Hi Joshua, Thanks for the reply. In this case we get a special SIP header in the 302, but I guess we'll need to find another solution to use it. On Wed, 27 Apr 2022 at 21:27, Joshua C. Colp wrote: > On Wed, Apr 27, 2022 at 5:33 AM David Cunningham < > dcunning...@voisonics.com> wr

Re: [asterisk-users] Context for 302 Moved response

Apr 2022 at 18:57, Jon Bonilla (Manwe) wrote: > El Wed, 27 Apr 2022 12:27:03 +1200 > David Cunningham escribió: > > > Hello, > > > > Does anyone know of a way to have a call go to a particular context when > a > > 302 Moved is received in response to

[asterisk-users] Context for 302 Moved response

the call somewhere different to all other calls. Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] T38 state values

Hi Joshua, Thank you for that. In the end it seems to have been a firewall blocking the UDPTL ports. On Thu, 24 Mar 2022 at 11:15, Joshua C. Colp wrote: > On Wed, Mar 23, 2022 at 7:07 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi Joshua, &

Re: [asterisk-users] T38 state values

again. On Fri, 18 Mar 2022 at 21:59, Joshua C. Colp wrote: > On Fri, Mar 18, 2022 at 12:27 AM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hello, >> >> We have a problem where one fax ATA connected to Asterisk works, and >> another ATA wi

[asterisk-users] T38 state values

nged to 3 on channel SIP/xx.xx.246.70:5060-0030e0a1 Notice the difference in the "T38 state changed to" values. Does anyone know what a value of 1, 2, or 3 means? I tried to find out from the Asterisk source code but it wasn't obvious. Thank you in advance for any tips. -- David Cunn

Re: [asterisk-users] Dahdi fails: fatal error: linux/pci-aspm.h: No such file or directory

Thanks Dovid. Installing from git does indeed work, I was wondering whether it had been released in a version and if so what version(s) that would be. On Tue, 22 Feb 2022 at 14:55, Dovid Bender wrote: > David, > > I vaguely remember having this issue on “newer” versions of Linux

[asterisk-users] Dahdi fails: fatal error: linux/pci-aspm.h: No such file or directory

Hello, I see some emails about a Dahdi compilation problem with "linux/pci-aspm.h: No such file or directory" two years ago, which suggest trying the "next" branch. Did this change go into a Dahdi release, and if so which version number(s) please? Thank you, -- David C

Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection

to add the features we need is what we're looking to hire someone for. Thanks. On Fri, 12 Nov 2021 at 13:20, David Cunningham wrote: > Hi Antony, > > Thanks for the suggestion. I didn't get a response on my request to join > the asterisk-dev mailing list. I'll try asterisk

Re: [asterisk-users] Willing to pay for patch to Asterisk fax detection

Hi Antony, Thanks for the suggestion. I didn't get a response on my request to join the asterisk-dev mailing list. I'll try asterisk-biz as well. On Fri, 12 Nov 2021 at 12:23, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Thursday 11 November 2021 at 22:29:

[asterisk-users] Willing to pay for patch to Asterisk fax detection

discuss pricing and your Asterisk development experience. If anyone has ideas for other places to advertise this request let me know! Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

Re: [asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10

//www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > ast

[asterisk-users] DAHDI compile failed in xusb_libusb.c

ols' Makefile:664: recipe for target 'all' failed make[1]: *** [all] Error 2 make[1]: Leaving directory '/usr/src/dahdi-linux-complete-3.0.0+3.0.0/tools' Makefile:9: recipe for target 'all' failed make: *** [all] Error 2 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1

Re: [asterisk-users] faxdetect timeout configuration

Hi Steve, Thanks for that. Perhaps the change to res_fax might help us? I'm hoping someone can say whether or not for sure. On Wed, 30 Dec 2020 at 11:00, Steve Edwards wrote: > On Wed, 30 Dec 2020, David Cunningham wrote: > > > Would anyone be able to tell us how to configure

[asterisk-users] faxdetect timeout configuration

this option for calls arriving via chan_sip? Is it just a matter of setting the FAXOPT(faxdetect) variable in the dialplan? What we'd like to do is restrict fax detection to the first N seconds of a call. Thanks very much for any advice, -- David Cunningham, Voisonics Limited http://voisonics.com

Re: [asterisk-users] NAT problem with recvonly calls

Hi Dovid, Thanks for that. Can you explain how the Progress() and/or Playback() actually help the NAT problem? I'm trying to figure out how it tells Asterisk the correct address to send the RTP to. On Thu, 3 Dec 2020 at 16:10, Dovid Bender wrote: > David, > > You should be able to do

Re: [asterisk-users] NAT problem with recvonly calls

Hi Dovid, We're using Enswitch so it uses AGI rather than a regular Asterisk dialplan, but perhaps sending it to a custom-made Asterisk context with the lines you suggest could be the best way forward. Thank you for that. On Thu, 3 Dec 2020 at 13:01, Dovid Bender wrote: > David, >

[asterisk-users] NAT problem with recvonly calls

this issue? Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

Thanks for the suggestions. We'd prefer not to complicate the architecture with additional proxies in front, so will try setting the Linux network routes to see if that helps. On Fri, 30 Oct 2020 at 16:24, John Runyon wrote: > David, can you play around with the routing table and get the

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

Bender wrote: > Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass > it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio > > On Thu, Oct 29, 2020 at 20:44 David Cunningham > wrote: > >> Hello, >> >> Does any

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the [general] section. Any suggestions would be greatly appreciated. On Sat, 24 Oct 2020 at 09:43, David Cunningham wrote: > OK, thank you George. > > > On Sat, 24 Oct 2020 at 03:16

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph wrote: > > > On Wed, Oct 21, 2020 at 9:16 PM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hello, >> >> We have an Asterisk server with two public IP addr

[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

TE sent from 2.2.2.2:5060 to pstn.com Does anyone know how this can be achieved? Thanks in advance for your help, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)

[asterisk-users] PJSIP - Forcing codec preference?

alaw) PS: I can't find the reference again but recall a recommendation to call Progress() due to nuances with some systems, is this still relevant? Regards David Herselman -- _ -- Bandwidth and Colocation Provided by ht

Re: [asterisk-users] Negotiates g729 but RTP contains g711

3918 (type 8, seq 020644, ts 000800, len 000160) [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020645, ts 000960, len 000160) Regards David Herselman From: asterisk-users mailto:asterisk-users-boun...@lists.digium.co

Re: [asterisk-users] Negotiates g729 but RTP contains g711

677e37c24e9> [2020-09-19 23:42:29] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 left 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9> [2020-09-19 23:42:29] VERBOSE[15153][C-00021a1f] pbx.c: Spawn extension (incoming, Downstream_01, 1

Re: [asterisk-users] Negotiates g729 but RTP contains g711

3918 (type 8, seq 020644, ts 000800, len 000160) [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020645, ts 000960, len 000160) Regards David Herselman -- ___

[asterisk-users] Negotiates g729 but RTP contains g711

receives g729 media from downstream iax2 trunk but then transmits g711a upstream. I'm however struggling with the downstream pcap, to establish what's different about these calls. Trunk config and forwarding structure works the identical way for 50+ other flows on the same host. Regards David

Re: [asterisk-users] rotatestrategy = none not working

Hi Steve, Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. On Wed, 20 May 2020 at 18:37, Steve Edwards wrote: > On Wed, 20 May 2020, David Cun

[asterisk-users] rotatestrategy = none not working

es the log files. Does anyone know why? Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] VoIP support engineer opportunity

have items you fit as well. 2. Provide your physical location, hours of availability, and indication of hourly rate. 3. Let us know what other work you have during business hours. Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28

Re: [asterisk-users] Hangup hook to put back a call into a queue

It might work for you to branch on ${DIALSTRING} just after your Dial command, if you want to handle a BUSY, NOANSWER, or other result. But if the peer of that Dial hungup, then based on what Joshua said, it seems there's no recovery. --

[asterisk-users] Looking for sample hangup_handler_pop and _wipe using vars

side. Cheers, David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https

[asterisk-users] Call disrupted...due to registration of third server?

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet

Re: [asterisk-users] Handling a non-responsive peer after it answers

Response below... On Fri, Dec 27, 2019 at 12:02 PM David P wrote: > > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > &

[asterisk-users] Handling a non-responsive peer after it answers

(args)) same => n,Dial(${AddressToReachPeer},2,b(${CONTEXT}^afterDialingPeerLogIpOfCb^1(${UUID}^${StartEpoch}))) same => n,Goto(handle${DIALSTATUS},1) Cheers, David -- _ -- Bandwidth and Colocation Provided by http://www.api-digi

[asterisk-users] One Touch Record and a matching entry in sip.conf

you in advance for any insight into this. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out th

[asterisk-users] Music on hold depending on who put call on hold

party puts the call on hold. Thanks in advance for any assistance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Find out which key ended recording?

Hi Steve, Thank you very much for that information. The result is the key in ascii perfectly! On Fri, 7 Jun 2019 at 18:05, Steve Edwards wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We're using Perl and so far I haven't found an equivalent there. > > On Thu,

Re: [asterisk-users] Find out which key ended recording?

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user

[asterisk-users] Find out which key ended recording?

recording. Note that only allowing # or * to end the recording won't work for us. Does anyone know how we can tell which key ended the recording? Thanks in advance for any help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28

Re: [asterisk-users] pjsip and tls client: How to decrypt Wireshark trace?

RIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Co

[asterisk-users] Change of H264 profile level problem

11.25.3. Thanks in advance for any assistance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Reliable information on which SIP party is transferring call

. Thanks in advance for any help. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

[asterisk-users] ChanSpy "Audiohook has stale audio in its factories" problem

to get 160 samples from write factory 0x1525f5d58ec8 [Jan 18 15:59:39] DEBUG[39503][C-0bbd] audiohook.c: Failed to get 160 samples from write factory 0x1525f5d58ec8 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

Re: [asterisk-users] CURL to post application/json

Thanks, Nasir, I'll see if that allows us to avoid SHELL. On Fri, 5 Oct 2018, 4:53 pm Nasir Iqbal, wrote: > Hi David, > > Have you tried CURLOPT function. > i.e > Set(CURLOPT(header)=Content-Type: application/json) > > Regards > > Nasir Iqbal > > ICTBroadcast -

[asterisk-users] CURL to post application/json

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is

Re: [asterisk-users] WebRTC as Softphone substitute ?

Thanks for sharing this, Alex. It sounds like TURN, as a media repeater, wouldn't work if the media must be secured (via SRTP). Is that right? On Wed, 3 Oct 2018, 3:17 pm alex epshteyn, wrote: > WebRTC requires a few specific things to be in place. We have blog posts > that talk about WebRTC

Re: [asterisk-users] How to know the IP of "manager show connected" in dialplan

How about using the CUT() function to get the IP address from the return from running the System() application running asterisk -rx "manager show connected"? I'm not in front of a machine, so cannot test this out... On Wed, 25 Jul 2018, 15:42 Ludovic Gasc, wrote: > Maybe I'm wrong, but, with

Re: [asterisk-users] How to steal an answered call?

Hello Patrick and others, Thanks, I wasn't familiar with the Bridge application and it may allow us to do what's needed. A transfer would of course be simpler but the user wants what the user wants... Thank you. On 9 July 2018 at 19:52, John Kiniston wrote: > David, > > You shoul

[asterisk-users] How to steal an answered call?

B, phone B answers the call, phone C dials something to "steal" the call from B, and finally A and C are talking. Searching on voip-info.org shows a "BristuffSteal" command but it's very out of date (Asterisk 1.2). Thanks in advance for any suggestions. Kind regards, -- David Cunn

Re: [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

, I get "Sorry, there was an error authorizing your account. Perhaps you did not approve authorization?" I've never received an email asking to verify my address, if that's what this error means. I just tried re-registering, too. On Sun, Jun 17, 2018 at 7:25 PM David P wrote: > I al

Re: [asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

I also just tried adding this: same => n,Set(SIP_CODEC_INBOUND=g722) On Sat, Jun 16, 2018 at 4:36 PM David P wrote: > We want to record inbound channels at 16kHz, but send only 8kHz to our > peers. I've set our default profile in sip.conf to disallow all but g722, > and the peers

[asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

. We're using Asterisk 14.7.6. Cheers, David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

FYI, we found that our peers don't hangup properly. But we would still like to know how to get the peer's hangup handler to fire upon peer hangup, because right now it corrupts our globals by firing after the caller's hangup handler. On Tue, Jun 5, 2018 at 5:40 PM, David P wrote: > FWIW

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

ose by extensions. No difference. On Tue, Jun 5, 2018 at 3:17 PM, David P wrote: > This has been super-helpful, Eric. However, the handleHangupByPeer priorities > below are still not run when the peer hangs-up. The last line in the cli > when the peer hangs-up is still: > Stri

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

then handleHangupByPeer runs. (And strangely, the value of global CB${IndexIntoPeers}CurrentCallsCount isn't accessible in handleHangupByPeer.) Cheers, David On Tue, Jun 5, 2018 at 12:58 PM, Eric Wieling wrote: > Don't use the _. pattern. Ever. > > The call has two channels so it ne

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

ed replacing the Dial above with: same => n,Dial(${DialForPeer},,g) Cheers, David On Tue, Jun 5, 2018 at 7:38 AM, Eric Wieling wrote: > Use hangup handlers, they work around the issues with the 'h' extension. > > On 06/05/2018 05:33 AM, David P wrote: > >> Thanks, Anthony. &

[asterisk-users] remove

-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

. Finally, is there a way to reset all globals, maybe as a variant of "dialplan reload"? On Tue, Jun 5, 2018 at 1:21 AM, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Tuesday 05 June 2018 at 08:33:26, David P wrote: > > > We're using Asterisk 14.7.6 and

[asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

g/asterisk-cmd-dial/ But it seems not to apply because I'm seeing the 'g' behavior without specifying that option, and the 'G' option seems intended for a far more complicated scenario. Cheers, David -- _ -- Bandwidth and Colocation Provide

[asterisk-users] Queue of automated members

o fallthrough, channel 'SIP/1000-0012' status is 'UNKNOWN' Cheers, David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.o

[asterisk-users] Long extensions that contain dashes

extension/destination_number in the call to Queue(). I couldn't find documention of any Queue() option like this. Is it possible to control the extension that the member receives? Cheers, David -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Getting DTMF from Asterisk Record?

. Thanks for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] Compiling 15.2.0 and 15.2.1 Fails Others are Fine

+ with all the latest updates. I've been using the rPi for about four or so years now and have not experienced a problem like this one. Any assistance will be greatly appreciated. Regards David. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Digium G100 and CID Dropping First Digit.

On Mon, Jan 15, 2018 at 10:41:27PM +, David Klaverstyn wrote: > port1 < Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) > port1 < Presentation: Presentation allowed of > network

[asterisk-users] Digium G100 and CID Dropping First Digit.

ng call from '21xx' to '049xx' on channel 0/5, span 1 Thanks David. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asteri

Re: [asterisk-users] Blocking outgping caller id on a PRI E1

It is likely being set by your PRI provider. Contact them to investigate. All the best... On 8 Nov 2017 9:03 am, "Neil Youngman" wrote: > I am trying to block/hide outgoing caller id on a PRI E1. > > It seems that it should be fairly simple, but it is defeating me.

Re: [asterisk-users] TDM400P takes too long to ring

info, or it is being supplied in a format that you have not set up for, this is likely the cause of the delay (looking for caller ID). All the best, David On 27 April 2017 at 12:48, Ryan, Travis <ry...@oscarwinski.com> wrote: > Hey all, > > > > I have a setup with tw

[asterisk-users] Advice of Charge for non-Snom SIP phones

t find any documentation to say what if anything is available. The "aoc_enable" setting doesn't seem to have any effect in sip.conf. Can anyone advise if there is any other support for AOC over SIP besides Snom, and how to configure it? Thank you, -- David Cunningham, Voisonics http://vo

[asterisk-users] Custom INFO for Advice Of Charge

with SIPSendCustomInfo but apparently it sends on all active SIP channels, and is only available with TEST_FRAMEWORK. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

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