Re: [asterisk-users] How to defer SDP in ACK for unit testing purposes
On Wed, Oct 10, 2018, at 4:27 AM, Olivier wrote: > Hello, > > I think I met a case similar to the one solved by [1] . Quoting this case : > > * res_pjsip: Handle deferred SDP hold/unhold properly. > > Some SIP devices indicate hold/unhold using deferred SDP reinvites. In > other words, they provide no SDP in the reinvite. > > A typical transaction that starts hold might look something like this: > > * Device sends reinvite with no SDP > * Asterisk sends 200 OK with SDP indicating sendrecv on streams. > * Device sends ACK with SDP indicating sendonly on streams. > > > Now, I would like to configure an Asterisk instance to act as those SIP > devices, ie to defer all SDP signaling in ACK. > > This is for testing purpose as I would like to reproduce in a lab an issue > with those SIP devices. > > 1. Is it possible ? I can use any Asterisk version for implementation. It is not possible to configure Asterisk for this. The chan_pjsip module only does normal reinvites with SDP when configured to pass through MOH signaling. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use AGi Commands without script in Dialplan
On Tue, Oct 9, 2018, at 1:52 PM, Yves wrote: > Am 08.10.2018 um 13:02 schrieb Antony Stone: > > On Monday 08 October 2018 at 12:44:43, Yves wrote: > > > >> I am looking for an easy way to execute any AGI Command directly from the > >> dialplan without the need to call an external script. > > The whole point of AGI is that it calls an external script in order to > > replace > > commands in the dialplan. > > > > Executing an AGI command without an external script doesn't make sense. > > > > > > Antony. > > > Hi Antony, > > thanks for your answer, even if it is a bit disappointing for me. I > understand the point... but... > why aren´t then all AGI-Commands also available as Dialplan Functions? > I can only find a small amount of functions for the dialplan that could > be seen as an equivalent > or near-equivalent of an AGI Command... I looked at the AGI command list and didn't see any that weren't possible in dialplan where it made sense. Do you have further examples? -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use AGi Commands without script in Dialplan
On Mon, Oct 8, 2018, at 7:44 AM, Yves wrote: > Hello, everybody, > often it is necessary to issue a single AGI command... > How can I realize this within a normal dialplan processing without > having to go the circumstantial way through an AGI script every time? > Why is it not possible to use the AGI commands like other functions > within the dialplan? > Although there are many dialplan functions that can be used as a > substitute for one or the other AGi command, or whose results are the > same, but not always... > > Example: > AGI_Command "Set Autohangup"... > > There is no way (at least of what I know) to set this AutoHangup feature > for a "normal" Call within the dialplan... and again, this is just an > example. I am looking > for an easy way to execute any AGI Command directly from the dialplan > without the need to call an external script. In particular for this it can done in dialplan using the TIMEOUT dialplan function[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_TIMEOUT -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
On Wed, Sep 26, 2018, at 6:46 AM, Olivier wrote: > Hello, > > This morning, I asked myself if WebRTC could be a viable alternative to > softphone deployment. > > For me, main issue with Softphones is the amount of work needed for > installation and configuration. > Also, Softphones must be carefully choosen if Deskphone-like quality is > expected. > > Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone > features (call history, BLF, ...) for WebRTC deployment simplicity. > > What do you think of this ? > What kind of experience did you met with such WebRTC deployments ? > What about classic telephony features (CallTransfer) ? > Have you tried Cyber Maga Phone 2K ? Speaking purely on CMP2K it's example code and is by no means a real phone. It was made to show off the video conferencing support. You'd be better off using JsSIP example code instead for making a solution in that area. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints
On Wed, Sep 26, 2018, at 10:25 AM, Floimair Florian wrote: > Hey all! > > I recently tried the dtmf_mode "auto_info" on my setup to support > endpoints that only understand SIP INFO as a fallback. > > My setup is the following: > > Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) > > Both are configured with "auto_info" dtmf_mode in pjsip.conf. > What I ran into is, that DTMF sent from endpoint A to endpoint B is > additionally sent via inband audio on the RTP stream from Asterisk to > endpoint B, as one can clearly hear the DTMF tone in the audio stream, > when a capture is played back on Wireshark. On the leg from endpoint A > to Asterisk there is no inband DTMF signal in the RTP audio stream. > > Can someone confirm this behavior? If yes than this is clearly a bug. > I had a look in the code which introduced this feature and couldn't find > anything obvious why this is happening. Have you bumped up the core debug to see what's going on underneath? There will be information about whether it is really generating the DTMF in the core, and if so then it'd be a result of the digit_begin function of chan_pjsip returning a value it shouldn't. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
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 calls and bridges with DAHDI/1-1. > >> > >> I send SIP/ to listen to a long, very long, file. > > > > Define "send". How are you doing it? > > > GoSub(play-long-file,s,1) You can't have a channel both in dialplan directly and also bridged to another channel at the same time. There's not enough context or information to really be able to answer without understanding fully. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
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. Define "send". How are you doing it? -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Community forum ?
On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > that forum supposed to supersede this mailing list ? Both remain available but the community forum seems to be more active, and it is easier to search and find things. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting invites to rtp ports ??
On Wed, Aug 29, 2018, at 10:34 AM, sean darcy wrote: > 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 > Call-ID: 1504207870-295758084-609228182 > CSeq: 1 INVITE > ... > WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on > 1504207870-295758084-609228182... > > I thought invites had to go to port 5060 or so. I don't understand why > somebody (let's assume a bad guy) is trying ports above 5. There is nothing that explicitly states that it has to be 5060, and in the case of the above it's just a random source port. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ContactStatus AMI Event on PJSIP Reregistration
Kia ora, I currently have a code review up[1] which removes the ContactStatus AMI event when a PJSIP endpoint re-registers. I removed it as I believed the ContactStatus event was redundant (as the status has not changed from the last event that would have been sent). Does anyone currently rely on this behavior? Cheers, [1] https://gerrit.asterisk.org/#/c/asterisk/+/9822/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird 'hairpin' call rtp audio problem
On Fri, Aug 10, 2018, at 11:10 AM, Benoit Panizzon wrote: > Hi Joshua > > > > The "rtp_keepalive" option can be used to have the RTP stack send an > > > RTP packet out. Try that and see what happens. > > > > Once again 'bullseye' that fixed the problem. Thank you! > > Now a customer with and FreePBX 2.9.0 (Asterisk 1.8.20.1) ran into the > same issue with our SBC. > > I told him to set rtpkeepalive=1 in sip.conf but I don't see this > version sending any comfort noise packets. > > Isn't there any way to disable this nat detection feature completely > in asterisk? (nat=no does not seem to do the trick) I don't remember 1.8 or the various options, you'd need to verify with the sample configuration file. Strict RTP protection can be disabled in rtp.conf using the strictrtp option. Otherwise it's not something in Asterisk that stops this kind of stuff, it's the NAT Implementation in the router. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings
On Thu, Aug 2, 2018, at 5:54 AM, Stefan Viljoen wrote: > Hi All > > With the below config, I just keep gettings this in the Asterisk 13.22.0 > CLI: > > WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type > registered for 'Agent' chan_agent doesn't exist anymore[1] in Asterisk 13 and above. [1] https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agent_pool/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
On Sat, Jul 28, 2018, at 6:28 PM, Jonathan H wrote: > OK, thanks. Shall I file a ticket to get that example file updated? Sure! -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
On Sat, Jul 28, 2018, at 6:27 PM, Jonathan H wrote: > Thanks, but... whoah! I think I just found a bug! > > As soon as I changed > accepts_registrations = yes > to > sends_registrations = yes > > and did a pjsip reload, Asterisk crashed. I tried starting asterisk. > Nothing. In the syslog: > > Jul 28 22:20:41 televox kernel: [ 50.728769] asterisk[1504]: > segfault at 0 ip 7f4be3e00646 sp 7ffc32067388 error 4 in > libc-2.27.so[7f4be3d4f000+1e7000] > Jul 28 22:22:02 televox kernel: [ 132.413114] asterisk[1579]: > segfault at 0 ip 7f62a9ba2646 sp 7ffc9215d408 error 4 in > libc-2.27.so[7f62a9af1000+1e7000] > > Took that line back out, and Asterisk started again. Shall I file a bug? Yes, issues should be filed on the issue tracker[1]. It may be something particular about your config. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote: > Using pjsip 2.7.2 on Asterisk 15.5 > Really struggling to make sense of translating these old 1.8 SIP > instructions into a neat pjsip_wizard conf suitable for 2018 > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 > > In pjsip_wizard.conf, I have the following, which seems to get me > registered, and it responds to an incoming call, but I always get > this: > > [Jul 28 18:32:29] NOTICE[22492]: res_pjsip/pjsip_distributor.c:659 > log_failed_request: Request 'INVITE' from '"demo" > ' failed for 'x.x.x.x:5060' (callid: > 5fa139428fef42d9bd0cd4063e10b047) - No matching endpoint found > > here's what I have in pjsip_wizard.conf > > [sip2sip] > type = wizard > sends_auth = yes > accepts_registrations = yes > transport = simpletrans > outound_auth/username = myusern...@sip2sip.info > outound_auth/password = password > remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info > endpoint/allow = alaw > endpoint/context = fromsip2sip > aor/max_contacts = 3 > registration/contact_user = myusername > outbound_proxy = proxy.sipthor.net > endpoint/language=en_GB This is an ITSP trunk, you've configured it kind of as if it were a phone. Instead of "accepts_registrations" you likely want "sends_registrations". Asterisk needs to register to them. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SRV with pjsip on Asterisk 15.5: yes or no?
On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote: > I'm trying to configure sip2sip, which says: > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk > "Asterisk, is currently unable to handle more that one result for a > DNS SRV lookup, and the Asterisk configuration needed for getting it > work with the SIP2SIP service is not trivial" > > It then gives a complex multi-section workaround in SIP. I remember > reading there'd be the same issue with PJSIP, and then I found this > post in the Asterisk blog from 2016: > https://blogs.asterisk.org/2016/04/20/pjsip-dns-support/ which says: > "chan_pjsip will now look for SRV records based on what transports are > configured on the system". > > Does this mean there's now a way of doing it? Because > https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip_wizard.conf.sample > says: > ; Hostnames must resolve to A, or CNAME records. > ; SRV records are not currently supported. > > H... I'm confused! SRV support for inbound matching was added after that comment was written. Identifying by IP address resolve a hostname down to all addresses (including SRV) - not just a single one. Outgoing supports A, , SRV, and NAPTR automatically. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-app-dev] how to use snoopChannel
On Tue, Jul 24, 2018, at 10:41 AM, Phil Mickelson wrote: > Ramesh, > > I'm also using ARI and Nodejs. I use the snoopChannelWithId command. I > always specify both directions so that the snooper can whisper to one of > the listeners and then just silence the snooper if I don't want anyone to > know they're listening. This is all done through the buffer that connects > the original parties. > > It's actually really simple and works very well. It's been in use in an > answering service for several years. I'm glad you've found it simple! It was a challenge to strike the right balance between the interface and giving full power over what you can do but I've found once it finally clicks people generally go "that makes sense". -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-app-dev] how to use snoopChannel
On Tue, Jul 24, 2018, at 10:20 AM, Ramesh C wrote: > Hello , > > I want to use spy and whisper using snoopChannel , but i do not understand > how to use this snoopChannel() method. > > Actually my need is that, one user on a channel wants to whisper/spy a talk > running between two caller(means both caller's channels are in a bridge), > so how i use this method please give me example code here. > and also give me possible value of arguments 'spy' and 'whisper' Who do you want to whisper to? Both channels? The way it works though is snoop channel creates a channel that you can treat like any other (add to a bridge, record, etc) - audio coming from it is a copy of the audio from the channel you are snooping on, audio going to it is whispered to the channel you are snooping on. To do spy/whisper you would snoop, create a bridge, and add the snoop channel and the channel doing the snooping into it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segfault on libasteriskpj.so.2
On Fri, Jul 20, 2018, at 3:18 PM, Carlos Chavez wrote: > I just finished installing a brand new server with CentOS 7.5 and > Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP > trunk) bridges with any SIP phone Asterisk crashes: > > Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfault at 188 ip > 7f158b9e047c sp 7f1568789820 error 4 in > libasteriskpj.so.2[7f158b976000+152000] > Jul 20 11:00:35 localhost kernel: asterisk[2925]: segfault at 188 ip > 7f44b1ea947c sp 7f700820 error 4 in > libasteriskpj.so.2[7f44b1e3f000+152000] > Jul 20 11:02:05 localhost kernel: asterisk[3133]: segfault at 188 ip > 7fe5b8f8147c sp 7fe547737820 error 4 in > libasteriskpj.so.2[7fe5b8f17000+152000] > Jul 20 11:08:25 localhost kernel: asterisk[3515]: segfault at 188 ip > 7f65c332547c sp 7f6551b56820 error 4 in > libasteriskpj.so.2[7f65c32bb000+152000] > > I found a closed bug: > https://issues.asterisk.org/jira/browse/ASTERISK-27210 but the status is > unresolved and closed. We are not using any DNS in our configurations, > everything is in IP notation. Internet is somewhat slow and sometimes > DNS fails the first time you need to resolve something. We started on > 13.21.1 but upgraded to 13.22.0 and are still having the same problem. > > This is not the first time I have seen this segfault. We tried a > virtual machine some months ago with the same result. At the time we > though the problem was related to the very old hypervisor we were using > but we get the same exact segfault on a new physical server. Anyone > else run into this bug? Any way to fix it or is it back to chan_sip? I'm not aware of any current issues for this and haven't seen any. I'd suggest filing an issue[1] with a backtrace so it can be narrowed down. It may be something particular to your usage that noone else has seen. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip realtime extensions
On Wed, Jul 18, 2018, at 7:23 PM, Benjamin Marty wrote: > Hello > > I'm currently using Asterisk 13 with the chan_sip sip driver. The > extensions are offloaded via realtime module to a MySQL database (via > ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords > and the other stuff from the standard Asterisk database schema. > > Now I want to mgirate to Asterisk 15 and in the same go migrate vom > chan_sip to pjsip. I already did the setup as described in the wiki > (Setting up PJSIP Realtime). > > Is there a way to keep the old database schema? Or at least to still have > SIP user + SIP password in the same table? Or is the only way to use the > new pjsip schema where you have to use the endpoint + auth table and > basically split the informations. You must use the provided PJSIP schema. There is no mechanism to do otherwise, outside of somehow creating something yourself. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy whisper
On Thu, Jul 5, 2018, at 7:37 PM, Patrick Wakano wrote: > Hello Asterisk list, > Hope you are all doing well! > > We are using the MixMonitor application to record the calls and under some > situations the call can be spied using ChanSpy with whisper enabled. > Sometimes the spying channel is a person who can interact in the call, and > some other times it is a sound file playing a message. The problem is that > for some reason the MixMonitor does not record whatever is injected via > whisper in the call. It records only the call itself... I've done some > research and apparently the Monitor application would be able to record > everything (but I didn't verify it myself) Anyway why MixMonitor can't? > Also does anyone have an idea on how to record everything in the same file? The implementation was never written to allow this and no feature added to control it. Someone would need to go into the code, define what needs to happen and how it can be controlled, and implement it (in regards to MixMonitor and ChanSpy). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not matching longest prefix with include
On Tue, Jun 26, 2018, at 8:31 PM, Dovid Bender wrote: > On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle wrote: > > > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > > > Doug, > > > > I tried that as well. Even with my dialplan looking like this: > > > > > > > > Ordering by includes works for me under Asterisk 11 and 13 The context always takes priority over includes. Includes are only examined if there are no matches in the current context. It's always worked this way. Ordering includes as such is one way to control that. As it is the matching is working as expected. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More testing
On Wed, May 23, 2018, at 12:20 PM, John Kiniston wrote: > I got excited when I saw 8 new messages on the Asterisk list-serve this > morning, What discussions must be happening I thought! > > You are a tease sir. Pfft, you should know that the community site[1] is the happenin' place being one of the people who respond there :P. [1] https://community.asterisk.org/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Codec negotiation
On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote: > On Fri, 11 May 2018, Joshua Colp wrote: > > >> In the above example, even though the INVITE/SDP says they prefer gsm > >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose > >> to use gsm or ulaw? > > > > Yes. > > > >> Can it be asymmetrical? They send gsm and I send ulaw? > > > > Technically, yes. In practice it's a bit iffy - specifically because > > some DSPs in devices won't allow it - they require a single codec be in > > use for each direction. > > So, Asterisk will defer it's choice of codec to match the codec it detects > in the incoming stream? It depends on the channel driver and configuration. The chan_sip module always matching outgoing codec to the incoming codec. The chan_pjsip module has an option to do that (which is on by default). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Codec negotiation
On Fri, May 11, 2018, at 10:07 AM, Steve Edwards wrote: > > So, without examining the RTP, you cannot tell which codec was actually > used? >From an Asterisk perspective "core show channel" will also show you what is >currently flowing. > In the above example, even though the INVITE/SDP says they prefer gsm over > ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose to use > gsm or ulaw? Yes. > Can it be asymmetrical? They send gsm and I send ulaw? Technically, yes. In practice it's a bit iffy - specifically because some DSPs in devices won't allow it - they require a single codec be in use for each direction. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files
On Tue, May 8, 2018, at 5:48 PM, Dovid Bender wrote: > Hi, > > It is my understanding that while Hebrew is supported by Asterisk the sound > files are not shipped with it as they are no longer being maintained. Can > anyone advise on what's needed to maintain a specific sound package? We are > considering to support Hebrew and possibly Yiddish. The actual submission process including what is required is documented on the wiki[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Sounds+Submission+Process -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tones in MixMonitor recording
On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote: > Thanks very much for the reply Joshua! > So I guess that setting dtmfmode=auto would be the safest choice in order > to strip out the DTMFs from the recording, right? > Cheers! It should work. Personally I prefer explicit configuring instead of having things just try to figure out what is in use. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tones in MixMonitor recording
On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > Hello list, > Hope you are all doing fine! > > I have stumbled over some piece of dialplan code in which apparently they > were trying to avoid recording the DTMF tones in the wav file. It is really > messy and I am not sure if this really works. So after a bit of research I > found this comment ( > https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is > said: > > *"Asterisk strips the DTMF from the audio stream when configured for > inband, so internal stuff can react to the DTMF and so the other side does > not hear the tone unless they are using inband (in which case it is > regenerated)"* > So my questions are, what are the cases in which Asterisk regenerates the > DTMFs? Does it cause the recording to have the tone as well, or is it only > transmitted to the other leg without being generated to the recording file? > Also, what if one or both legs are RFC2833? From my tests the RFC2833 > events never show up in the recording, but I just want to confirm that this > is always true. If properly configured then Asterisk will always strip and regenerate the DTMF tone. You have to purposely misconfigure things to cause it to not get stripped. IE: DTMF is actually inband but you configure it for RFC2833. Since Asterisk wouldn't be listening to the audio stream, it would go right through and get recorded. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on PJSIP's endpoint section in wiki
On Fri, Apr 27, 2018, at 11:13 AM, Olivier wrote: > Hello, > > I don't know if this list is the best place to ask such question but here > it is, anyway. > > In page [1], I can read in PJSIP's endpoint section configuration reference: > identify_by username,location Way(s) for Endpoint to be > identified > > Then clicking over identify_by text, you can read: > identify_by ... supported options are username, ... and auth_username > > How do yopu read it ? > I would expect the first line to written as: > dentify_by username,auth_username Way(s) for Endpoint to be > identified The wiki documentation hasn't been regenerated lately (it's in queue to be fixed). "username,auth_username" would be correct. There's also others[1] depending on version. [1] https://github.com/asterisk/asterisk/blob/13/configs/samples/pjsip.conf.sample#L633 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Explain PJSIP user matching within inbound SIP trunks
On Fri, Apr 27, 2018, at 7:00 AM, Olivier wrote: > Hello, > > I'm setting an Asterisk 13.14.1 box (Debian Stretch with packaged > Asterisk) to implement SIP trunking services ie to both trunk with carrier > trunks and IPBX trunks from various brands. > > For various reasons, I was inclined to implement this services with > pjsip_wizard.conf and I'm realizing I still have some remaining questions. > > For the moment, letting registration questions aside, which of the > following sentences is correct for Asterisk's PJSIP stack: > > 1. it would identify an incoming call only looking at From header ignoring > IP settings (both IP address and port), > 2. it would identify an incoming call only looking both at From header and > IP settings (both IP address and port), > 3. it would identify an incoming call only looking both at IP settings, > ignoring From header for identification but using it for other things > (setting CallerID, ...). It depends on configuration, but ultimately it can only be identified using a single endpoint identifier - so not in combination, thus by From OR IP. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP global section ignored in Asterisk 13.14.1
On Fri, Apr 27, 2018, at 9:57 AM, Olivier wrote: > Hello > > I've just discovered this [1] invaluable blog post (thank you very much > Richard for writing it) and its reference to PJSIP's > endpoint_identifier_order setting. > > On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I edited > a pjsip.conf file with the following content (and nothing more): Your version is also quite old, and changes/improvements/tweaks have been made since then to the option. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP global section ignored in Asterisk 13.14.1
On Fri, Apr 27, 2018, at 9:57 AM, Olivier wrote: > Hello > > I've just discovered this [1] invaluable blog post (thank you very much > Richard for writing it) and its reference to PJSIP's > endpoint_identifier_order setting. > > On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I edited > a pjsip.conf file with the following content (and nothing more): > [global] > endpoint_identifier_order=auth_username,ip,username > max_forwards=50 This is incomplete. You need to also have "type=global". -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: WebRTC tutorial
On Wed, Apr 25, 2018, at 12:40 AM, Bruce Ferrell wrote: > OK, I've gone back and refreshed myself; When I try to access cybermega > in /var/lib/asterisk/static-http at port 8088 the asterisk debug shows: > > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session. Top level > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/ > index.html > [Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] > has no handler > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. > status_code:404 When using static-http you have to have /static at the front so the path would be: /static/cyber/index.html > > When I serve it from apache, the web ui appears, but never connects. > > Using the firefox dev tools/console I see firefox can't establish a > connection the server at wss://:8089/ws > > The asterisk debug log shows: > > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session. Top level > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws > [Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no > handler > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. > status_code:404 > > Suggestions? Is there anything in the console at startup stating that stuff didn't load? The module which does websockets is res_http_websocket, and you can see if all that is needed is loaded using: "module show like websocket" on the CLI. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable blind and attended transfer during call
On Fri, Apr 13, 2018, at 6:09 PM, Andrzej Nowrot wrote: > Hi > > Is there a way to disable blind and attended transfer during a call. No, DTMF features are not call time configurable. They are only grabbed when the channel is first bridged, not as they are potentially used. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Fri, Apr 13, 2018, at 11:56 AM, Benjamin Marty wrote: > The current behaviour is that Earlymedia video isn't working when NAT's in > between are involved. The source/destination IP's are correct. So the > client is sending Early media video + Early media audio to the Asterisk > Server "in the cloud" and the Asterisk Server "in the cloud" is sending > both to the IP where the Client is located. But strangely just the Early > media audio is passing the NAT to the recipent. > > My guess is that the NAT traversal for Early media audio is fine, but the > one for Early media video not yet. Can you propably comprehend something in > that direction? Or can you guide me to the code part where Asterisk is > doing the Port change when a NAT is detected and the Client itself is > sending "fake" RTP Early media traffic to get a NAT Binding for incoming > RTP Early media traffic? The code is in res_rtp_asterisk[1]. It's not complex and despite the comment is not specific to video. Without logs showing where things are coming from and going I don't really have anything else to add. [1] https://github.com/asterisk/asterisk/blob/master/res/res_rtp_asterisk.c#L6140 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP error No auth credentials for realm(s) 'asterisk' in challenge
On Mon, Apr 16, 2018, at 12:47 PM, Administrator TOOTAI wrote: > Le 16/04/2018 à 16:52, Joshua Colp a écrit : > > On Mon, Apr 16, 2018, at 11:47 AM, Administrator TOOTAI wrote: > >> Hi all, > >> > >> we are trying to move our servers from chan_sip to chan_pjsip. At this > >> time no problems with phones, they all register fine and can place > >> calls. But for a trunk we face problem and can't place calls despite the > >> fact that registration is OK. What we get is: > >> > >> [2018-04-16 16:08:33] WARNING[18665]: > >> res_pjsip_outbound_authenticator_digest.c:178 > >> digest_create_request_with_auth_from_old: Endpoint: 'sip.xxx.tld': > >> Unable to create request with auth. No auth credentials for realm(s) > >> 'asterisk' in challenge. > > > > The remote side challenged for authentication but your endpoint has no > > "outbound_auth" configured, so chan_pjsip has no idea of how to > > authenticate. > > > > Thanks Joshua, that did it. We already tested a sort of by inserting > line = yes and endpoint = sip.xxx.tld in registration stanza but this > didn't work Outbound registration and outbound calling have no relation. The "line" option is strictly for INBOUND calls from what you have registered to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP error No auth credentials for realm(s) 'asterisk' in challenge
On Mon, Apr 16, 2018, at 11:47 AM, Administrator TOOTAI wrote: > Hi all, > > we are trying to move our servers from chan_sip to chan_pjsip. At this > time no problems with phones, they all register fine and can place > calls. But for a trunk we face problem and can't place calls despite the > fact that registration is OK. What we get is: > > [2018-04-16 16:08:33] WARNING[18665]: > res_pjsip_outbound_authenticator_digest.c:178 > digest_create_request_with_auth_from_old: Endpoint: 'sip.xxx.tld': > Unable to create request with auth. No auth credentials for realm(s) > 'asterisk' in challenge. The remote side challenged for authentication but your endpoint has no "outbound_auth" configured, so chan_pjsip has no idea of how to authenticate. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass through registration / proxy
On Wed, Apr 11, 2018, at 9:28 AM, Telium Technical Support wrote: > I’ve been tasked with building the whole thing in just Asterisk (as an > exercise). Trying to figure out how/if Asterisk alone can do thi. Asterisk is not a proxy, so it doesn't have functionality like this built in. There may be ways to sort of do such things, like listening to an AMI event for a successful inbound registration, updating configuration, reloading it, and causing an outbound registration to get sent. It's hackish at best. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As I stated previously in order for media to go to the source IP address and port, media has to be received from the endpoint. If this doesn't happen then you'll see exactly this behavior - we'll send to the IP address and port they told us. There's nothing that Asterisk itself can do in that instance, the endpoint has to send media or place the correct IP address and port in the messages. Was any media received from it? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote: > wohoo, so if I unterstand it correctly with that patch early media video > works over the Asterisk server? In other words the Asterisk server get's > able to (process/)forward the early media video stream with that patch? The patch forwards video while in an early media state before the call is answered and bridged, yes. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > My understanding based on Wireshark analysis is that the signaling works > (also the recipent phone is displaying the video frame before accepting the > call), also the calling phone send video (i see that also via Wireshark) > but the recipent phone doesn't get any video from the Asterisk before the > call. Ah yeah video, I forgot that it was a recent change to add support for it[1]. It's not yet in any release. [1] https://gerrit.asterisk.org/#/c/8398/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote: > Yes, media is flowing through Asterisk because both client's are behind > different NAT's. This doesn't answer the question of what is ACTUALLY happening in the scenario you describe which is very important. > Do I need to do something special in the Call Flow? Or anything additional > to the pjsip.conf? The "rtp_symmetric" option as you've used causes Asterisk to send media to the source of media, but it requires us to receive media. If we don't receive it then we send media to where they've told us to send it, which as I've mentioned can be wrong. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > Hello, > > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). > > Now I would like to get Early Media Video working between clients in > different NATed networks. The 183 signalling goes trough perfectly, but > asterisk doesn't forward the Early Media RTP stream from the caller to the > recipent. You would need to examine things specifically and see where media is flowing. Is the recipient behind NAT? If so then until we receive media from them (wich may or may not occur with early media) we may not have the correct target of media. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip trunk config question + DNS related error messages
On Thu, Mar 29, 2018, at 4:38 AM, Kevin Long wrote: > Greetings, > > > I am getting the following error (below) continually in my asterisk log, > related to qualify_frequency I believe. I am trying to use sip trunking > with the company flowroute. > > 3 questions if I may: > > 1) Is using qualify_frequency with a sip trunk a common or recommended > practice? I figured it would function as a keep-alive and keep the > ‘pjsip show endpoints’ availability data up-to-date if I wanted to check > on the health of the trunk. Sound right? > > 2) Any idea what this error means? Googling showed almost nothing except > one other post to this list for a bug that was fixed in 14 , I’m on 15.x > > 3) Any other recommendations for this trunking config? > > > Thanks very much ! Especially jcolp and gtjoseph for answering my > queries in the past, sorry if I don’t always respond again as I haven’t > actually figured out a good way to do that unless I am subscribed to > receiving all mails from the list. > > > > > [Mar 28 23:17:43] ERROR[4812]: res_pjsip.c:3770 endpt_send_request: > Error 320047 'No answer record in the DNS resp > onse (PJLIB_UTIL_EDNSNOANSWERREC)' sending OPTIONS request to endpoint > flowroute > > > [flowroute] > type=auth > auth_type=userpass > password=** > username=** > [flowroute] > type=aor > contact=sip:sip.flowroute.com:5060 Is there any reason you aren't just using sip:flowroute.com here? PJSIP does SRV resolution so that'll use SRV instead which I know works. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?
On Mon, Mar 19, 2018, at 12:53 PM, Benoit Panizzon wrote: > Hey List > > I sometimes use our asterisk server to do some debugging or other PBX > and SBC. > > Now we have a case where a PBX is replying an incomming invite with 180 > ringing immediately. It looks like the SBC does not accept this. > > According to my understanding of the RFC 3261 any provisional (aka > 1XX) reply should be good enough to make the sender stop re-sending > invites and accept this as a reply from the destination. > > So 100 trying would be option and a reply could also be directly 180 > ringing. Indeed. In practice though you want to stop the retransmission immediately and you usually don't know of the appropriate response yet so 100 is sent. > > So maybe some RFC specialist could tell me how this is exactly supposed > to work of if I maybe missed some other RFC more clear about that topic. > > To try to reproduce the problem with our SBC, is there a way to tell > the asterisk, preferably PJSIP, to directly answer with 180 ringing > without prior 100 trying? The PJSIP channel driver has no option or ability to do this. I do not recall if chan_sip does. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Originate
On Wed, Mar 14, 2018, at 12:58 PM, Dan Cropp wrote: > I am using AMI Originate to perform a new outbound call. > > The SIP Provider we send the call to wants us to pass the caller id of > the person we are calling for in the Contact header. > > For the AMI Originate, I pass the caller id information data in the > CallerID field. However, this is never being passed through the PJSIP > INVITE header > > Action: Originate > ActionID: S598 > Channel: PJSIP/133@1002 > Exten: createcall > Context: MyContext > Priority: 1 > Timeout: 6 > CallerID: CustomerName <## > > Variable: CALLERID(num- > pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 > Async: true > > Is there a setting that's required on the PJSIP endpoint to allow > overwriting the INVITE packet's Contact header? > Is there something else I am missing to perform this? > > Have a great day! Contact is never used for callerid. The only option available is contact_user on the endpoint to change the Contact username, that's it. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting the number of parked calls in a parking lot
On Tue, Feb 20, 2018, at 2:00 PM, Tech Support wrote: > All; > > With Asterisk 11, it was trivial to get the number of parked calls in > each parking lot simply by issuing the "parkedcalls show" command. However, > with Asterisk 13, things are done very differently and the "parkedcalls > show" command no longer exists. So my question is, how do I get that > information with Asterisk 13? There is still a CLI command to inspect the parking lot. It's "parking show ". -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving from res_sip to pjsip and simple bridge
On Thu, Feb 22, 2018, at 9:18 AM, Michele Pinassi wrote: > Hi all, > > on my old Asterisk 14.x box i use queue for some offices. For example, > in this scenario phone 5710 is ringing (after passing through a > queue...) and 5349 answer using REFER: > > -- SIP/5349-0072 answered Local/SIP-5710@MemberConnector-0031;2 > -- Local/SIP-5710@MemberConnector-0031;1 connected line has > changed. Saving it until answer for SIP/5002-006e > -- Local/SIP-5710@MemberConnector-0031;1 answered SIP/5002-006e > -- Channel SIP/5349-0072 joined 'simple_bridge' basic-bridge > > -- Channel Local/SIP-5710@MemberConnector-0031;2 joined > 'simple_bridge' basic-bridge > -- Stopped music on hold on SIP/5002-006e > -- Channel Local/SIP-5710@MemberConnector-0031;1 joined > 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> > -- Channel SIP/5002-006e joined 'simple_bridge' basic-bridge > <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> > > 0xa081718 -- Probation passed - setting RTP source address to > 172.20.xx.xx:60640 > > on new Asterisk 15.2 i decide to move to PJSIP but this functionality > don't work and, on REFER, call dropped. > > Maybe there's something needs to be enabled or checked ? I don't understand the specific scenario here you are referring to with the REFER. A call is answered using a 200 OK sent back by the called party. Can you clarify further? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compiling 15.2.0 and 15.2.1 Fails Others are Fine
On Tue, Feb 20, 2018, at 11:09 PM, David Klaverstyn wrote: > Hi All, > > When 15.2.0 was released I tried to upgrade as I do when new versions > are released but it failed to compile. I figured it may be a bug so I > waited for the next release but 15.2.1 also fails in the same location. > I can download, and compile 15.1.5 no problems at all. I'm not sure if > it is a 15.2.x problem or something else. > > When I compile the following occurs which I could not find and answer for. > > ./libasteriskpj.so: undefined reference to `initBcg729EncoderChannel' > ./libasteriskpj.so: undefined reference to `bcg729Decoder' > ./libasteriskpj.so: undefined reference to `bcg729Encoder' > ./libasteriskpj.so: undefined reference to `initBcg729DecoderChannel' > ./libasteriskpj.so: undefined reference to `closeBcg729EncoderChannel' > ./libasteriskpj.so: undefined reference to `closeBcg729DecoderChannel' > > > I am running this on a Raspberry Pi 3B, Raspbian 9.3 with Kernel 4.9.79- > v7+ 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. This has been fixed[1] in the branch and will be in the next normal release. You can pull down the minor change from the review if you want. It tells PJSIP not to build with support for that. [1] https://gerrit.asterisk.org/#/c/8193/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # converts to %23
On Mon, Feb 19, 2018, at 9:56 AM, Marcus Kvarsell wrote: > It is in the To: Header. Encoding is supposed to be done in that case. This became the default in a later version, specifically the "pedantic" option in chan_sip was changed to default to "yes" instead of "no". If you really don't want it you can change it to "no" in sip.conf -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get "SMS" messages (http) into Asterisk "sip messages"
On Mon, Feb 19, 2018, at 6:08 AM, Kevin Long wrote: > > Hello, > > We are building a shim to get SMS messages (which come in from twilio > via an http post to our python web app), forwarded on to the appropriate > SIP client registered to asterisk. > > The application receiving the “SMS” via HTTPS from twilio does not have > a SIP component. > > I am hoping there are different ways to get the message details into > Asterisk so that it can create a MESSAGE and send it to the local > endpoint. > > Does anyone know the best way to get this information into Asterisk? Can > I do it with AMI, AGI, a file queue ? > > Would love to hear from anyone who has implemented something like this. > Outbound is the easy part. How are you handling inbound SMS->SIP ? ARI can be used to send messages[1] or you can use file based origination[2] to send a message using the dialplan[3]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Endpoints+REST+API#Asterisk15EndpointsRESTAPI-sendMessageToEndpoint [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_MessageSend -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # converts to %23
On Mon, Feb 19, 2018, at 4:24 AM, Marcus Kvarsell wrote: > Hello, > > I have a broblem in asterisk 15 where an ami originate suddenly converts > 58#+46435345534 to 58%23+46435345534. This happenend when upgrading > asterisk 1.8 to 15. Could anyone help me with how to resolve this issue? You'll have to be more specific. Where do you see the %23? In SIP? As the extension trying to be executed in the dialplan? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > Thanks again for the hint. > Here is the output from asterisk. > > The call is coming on Audocodes gateway from: pstn- > > But asterisk display: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Why not loolking up "pstn-" in sip.conf? It found pstn- using 10.10.0.8:5060 - if the request always comes from the same IP address and port it has no other way built in to differentiate between the two except by matching based on username in the 'From' header. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >> "EndPoint Phone Number"
> >>
> >> Channel: 3phone number: pstn-
> >> Channel: 4phone number: pstn-9998
> >>
> >> When I am calling " pstn-" the port number "Channel:3" lights up but
> >> asterisk is showing that the call is coming on "pstn-9998"
> >>
> >> -- Executing . Answer("SIP/pstn-9998
> >>
> >> Asterisk should be showing "pstn-" (not pstn-9998)
> >> Where is this label coming from?
> >
> > It is from the SIP entry in sip.conf that it was matched against.
> >
>
> Thanks for the input.
>
> In sip.conf I have relevant entries.
>
> [pstn-] ; incoming/outgoing calls on FXO port
> type=friend
> secret=spa354
> username=voice-
> mailbox=622 ; just for audiocodes error complain
> host=dynamic
> canreinvite=no ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=1
> insecure=invite
>
> [pstn-9998]
> type=friend
> secret=158567
> username=fax-9998
> insecure=invite
> mailbox=622 ; just for audiocodes error complain
> host=dynamic
> canreinvite=no ; (dtmf not wroking correctly without this one)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=
>
> My asterisk registration is correct as well:
> sip show users
> Username Secret Accountcode Def.Context
> ACL Forcerport
> pstn-9998 158567 incoming
> No No
> pstn- spa354 incoming
> No No
>
> Caller display ID from PSTN on FXO ports are working OK.
> The [pstn-] is channel: 4
> The [pstn-9998] is channel: 3
>
> If the call on Audocode is lighting UP "channel:3" the sip.conf should
> associate that call with [pstn-] (and not [pstn-9998])
Not necessarily. You appear to be doing IP+port based matching. If requests
always come from the same source IP address and port, then it would match only
one. Turning on sip debug using "sip set debug on" and verbosity using "core
set debug 9" would give you more information about each packet (including where
it is from) and what was actually matched based on it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
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Re: [asterisk-users] incoming call label
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>
> IN audocodes setting I have:
> "EndPoint Phone Number"
>
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
>
> When I am calling " pstn-" the port number "Channel:3" lights up but
> asterisk is showing that the call is coming on "pstn-9998"
>
> -- Executing . Answer("SIP/pstn-9998
>
> Asterisk should be showing "pstn-" (not pstn-9998)
> Where is this label coming from?
It is from the SIP entry in sip.conf that it was matched against.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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Re: [asterisk-users] Weird 'hairpin' call rtp audio problem
On Fri, Feb 2, 2018, at 10:37 AM, Benoit Panizzon wrote: > Hello List > > Asterisk 13.14.1 in use with pjsip stack. > > On the remote side is a SBC which performs some 'nat' detection. I > suppose this means the SBC listens from where it is getting RTP data > and then replies to that ip. > > As long as the asterisk is initiating the call this is fine, the > asterisk start sending RTP to the media IP of the SBC and the SBC is > sending media back. > > Now I want to do a hairpin call, simulating call forward on no answer > (yes this is the situation I observed the problem first) > > So incoming AND outgoing calls are via SBC. > > exten => destination,1,Progress() > exten => same,n,Playtones(ring) > exten => same,n,Wait(5) > exten => same,n,Dial(PJSIP/sip:external@sbc) > > What I now observe when I dissect this call via Wireshark (and set rtp > debug on etc). > > Call to destination is established, up to the Wait(5) we have two way > RTP audio between the SBC and the Asterisk. > > The external destination picks up the call. From what I see the media > ip addresses and ports are correct, no direct media is attempted. So > asterisk should 'simple bridge' oder 'native bridge' the call localy. > > But for some reason, the asterisk server is NOT forwarding any rtp, nor > is the SBC forwarding any rtp it is getting from it's remote side which > is definitely sending rtp data. (yes I have access to the SBC and did > sniff both sides). > > I fear, that both, the asterisk side and the sbc side are attempting > the same kind of nat detection and do not forward rtp until they > receive any packets. > > I did probably try all possible permutations of: > > direct_media=no > rtp_symmetric=yes > force_rport=yes > > But still no audio. > > Any hints on how to force asterisk to send the first rtp packet? The "rtp_keepalive" option can be used to have the RTP stack send an RTP packet out. Try that and see what happens. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Cannot write OGG/Opus streams. Sorry" - any ideas?
On Sun, Jan 28, 2018, at 7:18 PM, Jonathan H wrote: > Oh. Ok, thanks for the quick response. Any idea of a workaround, or when it > might be implemented? It's just that most of the speech recognition > services now prefer opus. I can transcode with Sox, but that just puts > another (short) delay and process into pipeline. Not a complaint, just a > question. I have no timeframe on when such a thing would be done. It's not something that has been requested before to the best of my knowledge. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Cannot write OGG/Opus streams. Sorry" - any ideas?
On Sun, Jan 28, 2018, at 5:34 PM, Jonathan H wrote: > So as y'all know, with your help I managed to get Opus installed at last. Yay! > > With excitement, I wrote my dialplan, dialled in, and > > [Jan 28 21:30:11] ERROR[29977][C-001d]: format_ogg_opus.c:95 > ogg_opus_rewrite: Cannot write OGG/Opus streams. Sorry :( > [Jan 28 21:30:11] WARNING[29977][C-001d]: file.c:468 fn_wrapper: > Unable to rewrite format ogg_opus > > Any idea where I'm going wrong? I googled that error and can't find it > anywhere, nor can I find any notes about not actually being able to > *use* Opus once installed, so if anyone can point me to where I've > gone wrong I'd be most grateful! The opus support can only be used currently for reading files and for transcoding (for example one leg in g722 and the other in opus, or for conference mixing). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.19.0 Now Available
On Fri, Jan 12, 2018, at 3:02 PM, Binarus wrote: > Thanks for taking the time, but ... > > On 12.01.2018 12:04, Joshua Colp wrote: > > >> Could this be one of the rare cases where 13 and 15 needed security > >> fixes, but 14 didn't? > > > > These are normal bug fix releases, not security releases. As such 14 did > > not receive a release. > > > > Interesting. The announcements for 13.19.0 and 15.2.0 you have made here > both list all issues which have been fixed in the section "Bugs fixed in > this release". However, > > ASTERISK-27480 > ASTERISK-27452 > ASTERISK-27337 > ASTERISK-27319 > > seem to be security related (according to the short explanation texts in > the announcements) and have been fixed both in 15.2.0 and 13.19.0. > > I am wondering why 14 does not suffer from them, or -if it suffers from > them- why they are not considered security related there. > > I highly respect your work and don't want to steal your time since I > have probably seriously misunderstood something, but could you please > shortly explain what the string "Security: " (aka "(Security)" and with > other wordings) at the beginning of the short explanation text for an > issue exactly means? If you check those specific issues on JIRA you can see the specific releases they went into. They were also done in 14 as part of the past security releases so they were still fixed there. The script just may not have been run with the proper arguments to generate things correctly. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.19.0 Now Available
On Fri, Jan 12, 2018, at 2:51 AM, Binarus wrote: > On 11.01.2018 20:51, Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > > Asterisk 13.19.0. > > This release is available for immediate download at > > http://downloads.asterisk.org/pub/telephony/asterisk > > > > The release of Asterisk 13.19.0 resolves several issues reported by the > > community and would have not been possible without your participation. > > > > *Thank you!* > > Thank you very much for caring so much about security and bug fixes! > > But in this case, I am slightly worried. I saw the announcements for > version 13 and version 15, but no announcement for version 14 yet. The > website currently still offers 14.7.5 for download. > > Could this be one of the rare cases where 13 and 15 needed security > fixes, but 14 didn't? These are normal bug fix releases, not security releases. As such 14 did not receive a release. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 15.2.0 Now Available
On Thu, Jan 11, 2018, at 4:34 PM, Ira wrote: > Re: [asterisk-users] Asterisk 15.2.0 Now AvailableHello Asterisk, > > Thursday, January 11, 2018, 11:59:28 AM, you wrote: > > > The release of Asterisk 15.2.0 resolves several issues reported by the > community and would have not been possible without your participation. > For amusement I tried this again. I'm running an old version of 32 bit > CentOS, one that just went off support and when I run ./configure it > fails while trying to download pjproject. It looks like it says it's > going to try, fails, tries again and then gives up. Running with > NOISY_BUILD does not any useful information. Looks like it never tries > to execute the download command unless it executes it silently. > > It's not important, I'm perfectly happy running 14, but always try to > run the most current if I can. Can you please file an issue[1] with all the information? [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging ARI debug messages
On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote: > Hi there! > > Is there any way I can turn on debug for ARI and sending the output to a > separate log file? > So far I have only been able to turn on ARI debugging in the console > which results in the debug output being logged in /var/log/asterisk/ > messages > > I would love to have ARI debug log messages in /var/log/asterisk/debug > or even better in it's own ari-debug file. That is not something anyone has implemented as of this time. The messages themselves just get raised as normal verbose messages. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
On Tue, Jan 9, 2018, at 10:01 AM, Benoit Panizzon wrote: > Hi Jushua > > > The rtp_ipv6 option is not needed, in current versions things will > > automatically be updated to reflect the signaling. Remove it and give > > it a try. The option itself actually had the bug that you are seeing. > > Ok, commented out rtp_ipv6 in the config and did try again: > > IPv6 Registered client. > > c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80 > > Reply from * > > c=IN IP6 2001:4060:dead:beef::1 > > IPv4 registered client: > > c=IN IP4 157.161.4.172 > > Reply from * > > c=IN IP4 157.161.57.1 > > Perfect! It didn't occur to me to completely comment out that option as > I believed it was needed for rtp to work over ipv6. > > Thank you for that exceptional quick help. It used to be required but as part of Asterisk 14 work was done in DNS land (failover to different targets, including between IPv6 and IPv4) and based on discussions I had with other people at SIPit I made it automatic so that media family = signaling family. To keep things better in line and to provide a better experience the change was also done in Asterisk 13. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
On Tue, Jan 9, 2018, at 9:48 AM, Benoit Panizzon wrote: > Dear List > > I fear I stumbled over a bug in asterisk 13.14.1. > > My 'phones' are roaming around, sometimes some are connecting from ipv6 > enabled networks, another time they are not. > > If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat > problems. > > I have not specified a transport in the endpoint section, so that the > appropriate transport which corresponds to the registration can be used. > > Now I have noticed, if an phone is registered from an ipv4 only > endpoint and is performing an outgoing call, my asterisk server is > answering with an IP4 RTP IPv6 address: The rtp_ipv6 option is not needed, in current versions things will automatically be updated to reflect the signaling. Remove it and give it a try. The option itself actually had the bug that you are seeing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To Header instead of Request URI based routing
On Fri, Dec 22, 2017, at 9:54 AM, Benoit Panizzon wrote: > Dear List > > It looks like the common way to to sip signaling over a trunk is: > > In the Request URI, return the 'Register' Contact. > In the To: Header, send the destination number. > > Unfortunately, asterisk with pjsip (i did not try chan_sip) does > expect the dialed extension as request uri and does ignore what it is > getting in the To: header. > > I could not find any hint in the documentation of this can be changed. > > I found instructions for a work-around: > > http://www.kempgen.net/voip/sip-request-uri-vs-to-header-routing.html > > In the meantime: Is there a way to tell the asterisk with pjsip to use > the To: header to address an extension? Both chan_sip and chan_pjsip use the request URI, there's no configuration option currently to change it. Most people end up just doing the parsing in the dialplan. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?
On Mon, Dec 18, 2017, at 12:04 PM, Dan Cropp wrote: > Thanks George > > I originally didn’t have the 1002@ for the identify. Changed that when > things were not working. I changed it back. > > Unfortunately, the system I am connecting with doesn’t seem to support > the line support. Looking at the SIP packets, I see Asterisk send it. > Unfortunately, they do not send the line information as part of the > INVITE. I checked with some developers of that system and they do not > know anything about the line setting. > Is there any rfcs I could refer them to? "line" support doesn't have an explicit RFC. It relies on the remote side sending back the contents of the registered Contact address as they are supposed to when sending the INVITE. In practice some do, some don't. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP OPTIONS
On Wed, Dec 13, 2017, at 09:38 PM, volga...@networklab.ca wrote: > Hello Joshua, > What will be example of endpoint configuration that not require > authentication from specific ip ? An endpoint doesn't know about an IP address. The identify I previously mentioned is what associates the request to an endpoint. In the endpoint if you have no inbound authentication specified (auth option) then it won't require authentication. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".
On Sun, Dec 10, 2017, at 05:51 PM, Jonathan H wrote: > Hang on, all of the fiddling in this thread seems remarkably > over-complicating what should be an incredibly simple task. > > We know that a DTMF keypress interrupted the recording. We also know > that app_record.c knows which keypress it was from > > * \param dtmf_integer the integer value of the DTMF key received > > as in > > static enum dtmf_response record_dtmf_response(struct ast_channel > *chan, struct ast_flags *flags, int dtmf_integer, int terminator) > > For reasons which have me scratching my head, app_record turns a > useful DTMF value into a rather meaningless "DTMF" in the > RECORD_STATUS variable. When originally added it was only possible to terminate based on a termination DTMF, so you'd know which DTMF key was used because no other DTMF would stop. Afterwards a community member contributed a change[1] to add an option to allow any DTMF key to terminate it, but the dialplan variable stuff was not extended to make the knowledge of which DTMF was used available. > But SOMETHING must be floating around in Asterisk for app_record.c to > know what number was pushed. If I'm using RFC2833, is there ANY way of > getting that last keypress. > > In other words: "The user pressed a number, recording stopped, now > what was that number?" - WITHOUT rewriting and recompiling a core > application or doing any complex workaround? Within the code f->subclass.integer is where the DTMF digit is. You'd need to make a code change to set another dialplan variable which contains it. [1] https://issues.asterisk.org/jira/browse/ASTERISK-14380 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to read or write Geolocation (RFC6442) data in SIP/PJSIP messages ?
On Fri, Dec 8, 2017, at 10:58 AM, Jean Aunis wrote: > Hello, > > As far as I know there is no way to read or write the INVITE's body, > neither with chan_sip nor chan_pjsip. This is correct. There is nothing in Asterisk or the channel drivers to allow this. You would need to define a mechanism to do so and implement it in the code. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP OPTIONS
On Sun, Dec 3, 2017, at 10:55 AM, volga...@networklab.ca wrote: > If understand correctly type=identify is more for sip trunk > configuration ? > > > ;[mytrunk] > ;type=identify > ;endpoint=mytrunk > ;match=198.51.100.1 > ;match=198.51.100.2 > > > In chan_sip it was just reply 200 OK on keepalive packet without need > define trunks. > > All incoming traffic into chan_pjsip is matched to an endpoint, this includes OPTIONS. The OPTIONS request is also treated as if it were an INVITE per the RFC, which is why the extension also has to exist. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP OPTIONS
On Sun, Dec 3, 2017, at 10:42 AM, volga...@networklab.ca wrote: > Right now it reply 401 Unauthorized with message in log "No matching > endpoint ..." > on Content 0 should reply 200 OK I guess > > <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> > OPTIONS sip:10.30.100.27:5080 SIP/2.0 > Via: SIP/2.0/UDP > 10.30.100.41;branch=z9hG4bKf5eb.1ac76487.0 > To: > From: > <sip:vprx00@10.30.100.41>;tag=39e512215177543fc3584bac4ab8-64cb > CSeq: 10 OPTIONS > Call-ID: 66bf010933a080fe-17271@10.30.100.41 > Max-Forwards: 70 > Content-Length: 0 > User-Agent: kamailio (5.0.4 (x86_64/linux)) > > > [Dec 3 09:40:38] NOTICE[3511]: res_pjsip/pjsip_distributor.c:659 > log_failed_request: Request 'OPTIONS' from '<sip:vprx00@10.30.100.41>' > failed for '10.30.100.41:5060' (callid: > 66bf010933a080fe-17271@10.30.100.41) - No matching endpoint found You would need to add an endpoint for it and have it match, using a "type=identify" section matching on IP address would work. You would also need an "s" extension in the context since the OPTIONS request has not extension provided. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP OPTIONS
On Sun, Dec 3, 2017, at 10:34 AM, volga...@networklab.ca wrote: > Hello Everyone, > How to configure PJSIP to reply 200 OK from upstream sip proxy on > keepalive packet ? > > proxy ~> Keepalive OPTIONS ~> asterisk > <~ 200 OK <~ You would need to show the OPTIONS and what is happening now. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Trunk 401 Unauthorized (Alestra Mexico)
On Sat, Dec 2, 2017, at 06:33 PM, Carlos Chavez wrote: > I am having a really bad day trying to get incoming calls to work > on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where > everything was working but there seems that something got lost in > translation. No matter what I try I always get a 401 Unauthorized > message when receiving a call from the PSTN provider. I can make calls > and the registration is working. I have tried to set the identify to an > endpoint that does not have an auth defined. Anyone using Alestra SIP > trunks in Mexico? > > My identify is: > > = > endpoint : Alestra > match : 200.94.59.150/255.255.255.255 > match_header : > srv_lookups : true > > > It does not matter if I use the original endpoint or an endpoint with no > auth. Asterisk will still reject the call. Any tips? How can I make > sure that the identify is being used? If you turn up the core debug to level 4 and send it to the console it will tell you what it is doing. I'd also suggest providing the endpoint definition, and confirming it was loaded as expected. If it's not then you can look at the Asterisk console at load time and it will tell you what it did not like. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SOLVED! Re: pjsip subscribe (presence) always returns: No matching endpoint found
On Sat, Dec 2, 2017, at 07:53 AM, Benoit Panizzon wrote: > Hi List > > Just in case someone else runs into the same problem migrating from > chan_sip to res_pjsip. > > In chan sip you did define the voicemail variables in the peer section. > > I did configure most of that stuff into the endpoint of pjsip, > including: > > mailboxes= > voicemail_extension= > > Well, after re-reading where each config could be specified I found > that those two can also be used in an aor section. > > I moved them there, and now subscription to MWI works as expected with > my SNOM M9 phones, the 'hint' from the dial plan is being found. The "hint" isn't used for MWI, it's strictly used if you are subscribing to get the state of another extension. As you've discovered, though, the placement of mailboxes is important. When set in an endpoint it configures unsolicited MWI - that is MWI without the endpoint subscribing for it. When set in an AOR it configures what mailboxes the endpoint can subscribe and receive MWI for. Since you've moved it to the AOR it can now subscribe to the mailbox and receive MWI. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip Transfer 'Failed to parse destination uri'
On Mon, Nov 27, 2017, at 05:42 AM, Benoit Panizzon wrote:
> Ok, answering myself:
>
> Asterisk 13.14.1~dfsg-2+deb9u2
>
> Apparently suffers the pjsip transfer bug described @
>
> https://reviewboard.asterisk.org/r/4316/diff/
>
> Specifying the full URI:
>
> Transfer(PJSIP/sip:${DESTEXTEN}@trunk) does resolve the URI parsing
> problem and is sending back the 302 message (which does not containg a
> Diversion header, Jay promising, testing that next), but as described in
> the Bug the Contact header is being messed up.
PJSIP requires a full SIP URI that can be used by the remote endpoint to
be provided. The code does not look up an endpoint and try to construct
a SIP URI for you.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] pjsip multiple transports for one endpoint (dual stack) ipv6
On Sat, Nov 25, 2017, at 03:38 PM, Benoit Panizzon wrote: > Hi List > > I have stumbled over the next question google didn't answer. > > I have a dual-stack environment, ipv6 and ipv4. > > With chan_sip asterisk was listening on ipv6 and ipv4 simultaneously. > > I did try to define to have pjsip listen to the ipv6 address including > ipv6 mapped ipv4 addresses: > > [transport-udp] > type=transport > protocol=udp > bind=[::]:5061 > > Google told me, pjsip is not able to parse the ipv6 mapped ipv6 > address, you have to listen to ipv4 separately. > > So next try: > > [transport-udp] > type=transport > protocol=udp > bind=[::]:5061 > bind=0.0.0.0:5061 > > Well, this also does not work, only one address family is being bound. > > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0:5061 > > [transport-udp6] > type=transport > protocol=udp > bind=[::]:5061 > > [phone] > type=endpoint > transport=transport-udp > transport=transport-udp6 > > Well, this sort of works, the phone is able to register, but qualify > does not work, as it seems to always try to use the 2nd transport > definition. > > So how the heck do you define a endpoint listening to both ip versions > if you don't know if your roaming phone is going to connect from a ipv6 > enable VoLTE Network without NAT problems, or is going to use the old > legacy ipv4 protocol? :-) Don't specify a transport on the endpoint. Transport selection will automatically choose the right one in this scenario. The "transport" option only allows a single transport and it is for forcing a transport to always be used regardless. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip subscribe (presence) always returns: No matching endpoint found
On Sun, Nov 19, 2017, at 02:11 PM, Benoit Panizzon wrote: > Hi Joshua > > thank you for the quick reply > > > Have you checked the Asterisk console when PJSIP is loaded to see if > > the endpoint did not load for some reason? Does it show up in "pjsip > > show endpoints"? > > Yes, the endpoint shows up. > > Endpoint: 11/(scrubbed from mail) > Not in use0 of inf > InAuth: 11/11 > Aor: 11 1 > Contact: 11/sip:11@[2001:4060:dead:d1d0:204:13ff:fe 58af7d6822 > Avail 5.799 > Transport: transport-udp udp 0 0 [::]:5061 > > I had the qualify statement at the wrong place, but that's sorted out > now. > > But still, subscribing to the hint results in a 404 error. > > Acutualy, that subscribing is a bit odd, it's a snom M9 phone that is > trying to subscribe to itself. > That does not make much sense in my opinion. > > It just that chan_sip reported OK to this and chan_pjsip replies with > 404. > Or is pjsip more intelligent and trying to prevent the phone from > subscribing to itself? The chan_pjsip module doesn't prevent that. You'd need to provide the full SUBSCRIBE now that it is actually finding the endpoint and coming in. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip subscribe (presence) always returns: No matching endpoint found
On Sun, Nov 19, 2017, at 12:11 PM, Benoit Panizzon wrote: > Hello List > > I am in the progress of migrating from chan_sip to pjsip. > > I fear I have missed something on how hints need to be specified for > pjsip. > > For chan_sip I have configured sip.conf > > subscribecontext = localuser > > > and in the dialplan I set: > > [localuser] > exten => 11,hint,SIP/11 > > Now if a phone subscribes to '11' this works. > > Now I try to get the same working for pjsip. I understood that for > pjsip the hit needs to be placed in the same context as the endpoint: > > [11] > type=endpoint > transport=transport-udp > context=localuser > disallow=all > allow=g722 > allow=alaw > allow=gsm > auth=11 > aors=11 > callerid=(remove in this example > qualify_frequency=10 > mailboxes=11 > voicemail_extension=411 > > And in the dialplan I changed: > > [localuser] > exten => 11,hint,PJSIP/11 > > But I constantly get: > > Request 'SUBSCRIBE' from '"Benoît Panizzon PJSIP" <sip:1...@woody.ch>' > failed for '2001:4060:dead:d1d0:204:13ff:fe30:228d:2332' (callid: > ow21f3eg@snom) - No matching endpoint found > > And I in the logger I see that the subscriber request is being rejected > with error 404. > > Any hints what I'm doing wrong? Have you checked the Asterisk console when PJSIP is loaded to see if the endpoint did not load for some reason? Does it show up in "pjsip show endpoints"? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SSRC =0x0 in RTP
On Wed, Nov 15, 2017, at 01:37 PM, Harel Cohen wrote: > > > > Hi Joshua, > > > Thank you for looking into this. > > > Their response IS based on traces I've sent them. Attached is such trace > in text format (server IP has been changed to 111.111.111.111). Some > repeating RTP packets has been truncated. > You can see that after the 200 OK SSRC is sent from the server to the > phone > as '0x0'. The same has happened with G729 codec. > > > Let me know if you need the full trace or anything else from my side. > > > I should also mention that this is Asterisk version 1.8.12.1 I'm sorry but this version is old enough that what I currently know is far past it. It may have been possible in that old version for the SSRC to be as you state. In recent stuff it doesn't seem to be possible. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Wed, Nov 15, 2017, at 01:30 PM, Carlos Chavez wrote: > Here is more information from the browser about the session: > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF > > On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the > endpoint. I have configured a STUN server in both rtp.conf and > res_stun_monitor.conf What is the exact network topology? Is Asterisk behind NAT as well with ports forwarded? If so you should configure a mapping in rtp.conf so that the internal IP address is mapped to its external IP address in the ICE candidates, giving a better chance that things will work. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote:
> On 11/14/17 5:23 PM, Joshua Colp wrote:
>
> > On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote:
> >> Trace with 3 clients. We can hear each other but no video.
> >>
> >> https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz
> > Do you see anything in the Javascript console of the browser? We are
> > adding the needed media streams by sending a reinvite to the
> > participants but we don't get any response, which means for some reason
> > the browser may not have liked what we provided.
> >
> This is what I get on the console:
> new session - outgoing - [object Object]
> cyber_mega_phone.js:78:3
> ontrack: audio - 8b7fca5e-bb67-4e8c-8bdb-84fb80ac4cc0 stream
> 66e4250b-c196-4482-a347-d12772ef865d
> cyber_mega_phone.js:111:4
> Streams: added 66e4250b-c196-4482-a347-d12772ef865d
> cyber_mega_phone.js:225:3
> ontrack: video - ad836e20-c0c9-423f-9c42-0aef19c5ca32 stream
> 66e4250b-c196-4482-a347-d12772ef865d
> cyber_mega_phone.js:111:4
> confirmed: adding local stream {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
> cyber_mega_phone.js:84:5
> Streams: added {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
> cyber_mega_phone.js:225:3
> RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use
> RTCPeerConnection.getSenders/getReceivers instead.
> cyber_mega_phone.js:82:17
> ICE failed, add a STUN server and see about:webrtc for more details
Looks like for some reason it failed to successfully do ICE negotiation
potentially on the newly added remote streams. Why that is is
environment specific - but the problem does seem to be on the web
browser/client side, not in Asterisk itself. You'd need to figure out
why.
This is one of the annoyances of WebRTC - the browser can be a black box
at time and when things go wrong (like this) it's hard to dig and figure
out what is up.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: > Trace with 3 clients. We can hear each other but no video. > > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz Do you see anything in the Javascript console of the browser? We are adding the needed media streams by sending a reinvite to the participants but we don't get any response, which means for some reason the browser may not have liked what we provided. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: > On 11/14/17 4:27 PM, Joshua Colp wrote: > > > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: > >> On 11/14/17 3:55 PM, Joshua Colp wrote: > >> > >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > >>>>I followed the blog post and I can get video from the conference > >>>> if > >>>> I configure the bridge as follow_talker so I know everything is working > >>>> on the pjsip side. The only problem is that video_mode = sfu is > >>>> apparently not valid in either confbridge.conf or via the dialplan and I > >>>> get no video with that option. > >>> The option, when set, will show up as "no video" if you do the > >>> "confbridge show" as you mentioned. That's a bug which is why I > >>> mentioned filing an issue. It is still valid though. > >>> > >>> Have you confirmed that the maximum number of streams is set using > >>> "pjsip show endpoint"? and that the codecs are correct? > >>> > >> allow : (ulaw|vp8|h264) > >> max_audio_streams : 10 > >> max_video_streams : 10 > >> > >> Video and audio work fine if I use follow_talker in the > >> confbridge. No video when set to sfu. > > What browser are you trying from? Can you provide a SIP trace (pjsip set > > logger on)? And what is the output of "core show channel" for each > > channel when they are in the video conference bridge? > > > We have tried with Firefox (56) and Chrome 61.0.3163.100 on both > Windows and OSX. > > SIP trace: > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/GsXHb9EoRUZuJrZ > Channels: > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/9W04VCUFQSfVumW > > It appears that the CBAnn channels only have audio a no video. Those are the announcer channels for playing audio into the conference bridge. I need to see an attempt with more than 1 participant in the bridge. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: > On 11/14/17 3:55 PM, Joshua Colp wrote: > > > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > >> I followed the blog post and I can get video from the conference if > >> I configure the bridge as follow_talker so I know everything is working > >> on the pjsip side. The only problem is that video_mode = sfu is > >> apparently not valid in either confbridge.conf or via the dialplan and I > >> get no video with that option. > > The option, when set, will show up as "no video" if you do the > > "confbridge show" as you mentioned. That's a bug which is why I > > mentioned filing an issue. It is still valid though. > > > > Have you confirmed that the maximum number of streams is set using > > "pjsip show endpoint"? and that the codecs are correct? > > > allow : (ulaw|vp8|h264) > max_audio_streams : 10 > max_video_streams : 10 > > Video and audio work fine if I use follow_talker in the > confbridge. No video when set to sfu. What browser are you trying from? Can you provide a SIP trace (pjsip set logger on)? And what is the output of "core show channel" for each channel when they are in the video conference bridge? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > I followed the blog post and I can get video from the conference if > I configure the bridge as follow_talker so I know everything is working > on the pjsip side. The only problem is that video_mode = sfu is > apparently not valid in either confbridge.conf or via the dialplan and I > get no video with that option. The option, when set, will show up as "no video" if you do the "confbridge show" as you mentioned. That's a bug which is why I mentioned filing an issue. It is still valid though. Have you confirmed that the maximum number of streams is set using "pjsip show endpoint"? and that the codecs are correct? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge SFU for Asterisk 15
On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: > I am trying to get the "Mega Phone" demo working on my office PBX > but there seems to be a problem when trying to set the default bridge to > sfu mode. I have the following configuration in confbridge.conf in the > default_bridge section: video_mode = sfu but when I do a "confbridge > show profile bridge default_bridge" I see: > > Video Mode: no video > > I can change it to follow_talker, last_marked or first_marked and > it does change, it is just the sfu option that does not seem to be > valid. I am using Asterisk 15.1.2 for my testing. I even tried to > force the option via Dialplan: > > [ Context 'ext' created by 'pbx_config' ] >'1000' => 1. Answer() > [extensions.conf:0] > 2. Set(CONFBRIDGE(bridge,video_mode)=sfu) > [extensions.conf:0] > 3. ConfBridge(guest) [extensions.conf:0] > 4. Hangup() [extensions.conf:0] > > But I get no video at all on the conference. > > Any ideas? The CLI command doesn't have sfu as a mapping right now. Please file an issue[1] for that. As for video there's additional configuration that you have to do in pjsip.conf in order to allow the streams. I'd suggest following the blog post[2] which has known working configuration. [1] https://issues.asterisk.org/jira [2] http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SSRC =0x0 in RTP
On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote: > Hello, > I have a problem where on an outgoing call a Grandstream phone (GXP2130) > closes the incoming voice stream about 1 second into the call (the remote > party hears the Grandstream, the Grandstream doesn't hear thr remote > party). I have verified with logs and traces that this is not a NAT issue > or any other network-related problem. All incoming RTP packets arrive at > the phone on the correct port etc. as declared in the SDP. > I opened a ticket with Grandstream and they replied: " > > *the phone starts receiving RTP with SSRC =0x0 which is wrong".* > > Is this an Asterisk problem or the phones? Is this something that can be > fixed on the Asterisk side? Asterisk would be sending the RTP to the Grandstream. I'd suggest getting a packet capture using tcpdump or wireshark to confirm what they've said though. I just looked at the code and I don't see a way that we'd ever have the SSRC be 0. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to log missing RTP packets ?
On Fri, Nov 10, 2017, at 10:34 AM, Olivier wrote: > Hello, > > When a call is starting, Asterisk starts sending and receiving RTP > packets. > Each packet has a sequence number. > > 1. Instead of logging everything as rtp set debug is currently doing, is > there a way to only log: > - the sequence number of the first received packet, > - any missing or misplaced sequence number. > > 2. Is there a way to log RTP debug information in a specific file or send > the same date to a custom daemon or filter ? There's nothing built into Asterisk itself to do this already. Logging to something like Homer might work, or just doing a packet capture. Otherwise you'd need to make changes to Asterisk to add the functionality you mention. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
On Mon, Nov 6, 2017, at 02:14 PM, Saint Michael wrote: > Asterisk is unique in terms that we can create new applications that talk > to databases and generate any logic whatsoever. Asterisk is a development > environment for anything telecom, not a PBX. I believe that we need to > make > PJSIP more efficient so Asterisk can expand its footprint. > Please tell somebody to add a way to prohibit PJSIP from proxying RTP. I > can help if you give me some directions, but I understand the complexity > of > PJSIP under the hood. There is noone to be "told" to do such work. Asterisk is an open source project that includes not just Digium but also other contributors. It's all of us working together that helps to improve Asterisk. Like everything in life we all have our own priorities and responsibilities, which differ, that drive what we all work on and have an interest in. As for providing direction for doing the work yourself - this is not something that anyone has looked into or planned so there's no real direction to give. You have to pick somewhere to start, dig, and figure out what needs to be done. If you have specific questions then the asterisk-dev mailing list would be the best place to discuss such things since that is where developer talk occurs. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
On Sun, Nov 5, 2017, at 02:42 PM, Saint Michael wrote: > Now that Joshua had the kindness to respond, I see here a big disconnect > between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like > me > would need to avoid proxying the media. Would would Digium support and > bring in with such fanfare a channel like PJSIP that lacks the only thing > that 99% would need to do business in an efficient manner? I mean people > like me buy and sale billion of minutes every day, and most of my peers > gravitate towards Opensips and other solution that do not touch the > media. > Yesterday I had to roll back my sleeves and go back to the old sip > channel. > I would love to see Asterisk-PJSIP to find a way to act like a proxy. > This > would turn Asterisk into a real wholesale business tool, which is not, so > far. It's not the lack of this feature which drives people to using OpenSIPS or Kamailio for this use case. It's just fundamentally designed differently and better performant for that scenario. Asterisk isn't the best solution for everything everyone needs or wants, and that's okay. There are other projects (like those already mentioned) that are a better fit, and Asterisk can even play a part in there as an application server. I'm a firm believer in using the right tool for the right job even if it means that Asterisk isn't the right fit. Frustrated users are something I never want to see. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP and Non Media Proxy
On Sun, Nov 5, 2017, at 07:16 AM, Saint Michael wrote: > Please correct me if I am wrong. With PJSIP there is no way for Asterisk > to > stay a OUT of the media path, while with the old SIP channel, using > directrtpsetup and directmedia, it just works. The issue I think is that > other servers do not accept reinvites or updates to redirect media, so > PJSIP will not be able to step out ever. Using the old sip channel, the > 200 > OK with SDP tells the calling side to talk direcly to the other side. > Is there a way to do this with PJSIP? There is no "directrtpsetup" equivalent in PJSIP. Even in chan_sip it was experimental and could break things depending on the codec payloads in use. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip insecure=port,invite
On Thu, Nov 2, 2017, at 04:50 AM, Dmitry Melekhov wrote: > Hello! > > Looks like faq, but... > > Could you , please, point me on how to convert this > > > [cisco] > type=friend > host=192.168.22.253 > insecure=port,invite > > > to pjsip? The equivalent is this: [mytrunk] type=identify endpoint=mytrunk match=203.0.113.1 >From the page you linked. That says "Match incoming traffic from 203.0.113.1 and use endpoint mytrunk for it". You also need an endpoint defined like: [mytrunk] type=endpoint context=from-external disallow=all allow=ulaw -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP trunk to Telynx
On Fri, Oct 20, 2017, at 10:17 PM, Carlos Chavez wrote: > Has anyone used Telynx as a SIP trunk provider? It works with chan_sip > but it I seem to be having problems trying to set up a PJSIP trunk. I > always get a 401 Unauthorized when they send me a call. I know my > username and password are correct since I can register and PJSIP uses > the same information for inbound as for the registration. Unfortunately > their support department said "PJSIP what?". It seems mos SIP providers > know Asterisk but are not aware of the important change coming. I > already got a nasty surprise from Voicepulse stating that they do not > support PJSIP so their service will not work with newer installations. Generally ITSPs don't authenticate to you, they expect the device or software to just know the call is from them and to accept it. In PJSIP this is done by using an identify section and matching based on IP address. There's also the line option[1] to outbound registration which works with some equipment, if it works then no identify section is required. [1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP add header not working
On Mon, Oct 2, 2017, at 12:06 PM, Andre Gronwald wrote:
> Hi,
> I am trying to add a custom header to my calls to map several call-legs
> into a global call for viewing.
>
> For this to work I read the call-id from pjsip-channel and write it into
> X-CID:
>
> ##
> -- Executing [s@macro-dialout-trunk-predial-hook:4]
> Set("PJSIP/10-0006",
> "pjsipCallId=313530363933383438363436353930-1gh0bjceo933") in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:5]
> Set("PJSIP/10-0006",
> "PJSIP_HEADER(add,X-CID)=313530363933383438363436353930-1gh0bjceo933")
> in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/10-0006",
> "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(num,i)=0xx)") in new stack
> -- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/10-0006",
> "1?Set(CONNECTEDLINE(name,i)=CID:3x)") in new stack
> -- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/10-0006",
> "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3x)") in new stack
> -- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/10-0006",
> "0?customtrunk") in new stack
> -- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/10-0006",
> "PJSIP/0xx@3x,300,T") in new stack
> -- Called PJSIP/0xx@3x
The PJSIP_HEADER dialplan function operates on the channel it is invoked
on. In this case you are using it on the caller, not the called party.
The wiki documentation[1] includes an example of how to apply it to an
outgoing call.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Gerrit usage?
On Fri, Sep 29, 2017, at 12:16 PM, Daniel Tryba wrote: > I'm trying to figure out how to commit some code for review. Following: > https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage > > Created a ssh alias. > Cloned using: "git clone ssh://asterisk/asterisk" > Set name and email. > Installed the gerrit commit hook: "git review -s" > Try to change to asterisk 13 for creating a patch: "git checkout 13" > This fails with: > error: pathspec '13' did not match any file(s) known to git. > > > "git checkout -b 13" appears to fix this. This did not create a branch from 13. This created a branch named "13" from the branch you were on, which was most likely master. That is why your "git review" is not working as you expect, because you are telling it that you did the work against "13" but it really was against master. git checkout -b 13 origin/13 Would create a local branch "13" which is from the remote branch "13". You'll need to do this, or do your "git review" against master and then cherry pick from inside Gerrit to the appropriate branches. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
On Tue, Sep 26, 2017, at 05:53 PM, marek cervenka wrote: > Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): > > On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > >> hi, > >> > >> i want use asterisk+pjsip as voip client with multiple registrations > >> (perf testing) > >> > >> i'm using this example configuration for one account > >> > >> [308] > >> type=registration > >> outbound_auth=308 > >> server_uri=sip:3...@example.com:5060 > >> client_uri=sip:3...@example.com:5060 > >> > >> [308](auth-userpass) > >> username=308 > >> password=pass > >> > >> [308](aor-single-reg) > >> contact=sip:example.com:5060 > >> > >> [308](endpoint-basic) > >> outbound_auth=308 > >> aors=308 > >> > >> [308] > >> type=identify > >> endpoint=308 > >> match=example.com > >> > >> > >> my problem is contact on the other side (is same for all endpoints) > >> > >> Addr->IP : 1.1.1.1:5060 > >> Reg. Contact : sip:s@1.1.1.1:5060 > >> > >> all incoming calls from PBX to my Asterisk are routed to only one > >> account (because of same ip address/port ?) > >> > >> how can i specify different source port or different contact address for > >> asterisk pjsip client with registration? > > The "contact_user" option configures the user portion of the Contact > > that is sent in the REGISTER. You can set it to a different value for > > each registration. > > ok i have this configuration now > client - asterisk+pjsip (public ip 1.1.1.1) > pjsip/307 > pjsip/308 > > server - asterisk+chan_sip (public ip 2.2.2.2) > sip/307 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:307@1.1.1.1:5060 > > sip/308 > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:308@1.1.1.1:5060 > > > now, every call from server to client is received through pjsip/307 . > but i need receive call for pjsip/308 through registration of pjsip/308. > is it possible? > is it possible configure different source port other than 5060? There is no ability to match to an endpoint currently based on the transport traffic comes in on. You can try enabling the line option[1] which may allow the inbound calls to be directed to an alternative endpoint. If this doesn't work you'll need to match all incoming to a single endpoint and direct things appropriately in the dialplan based on the dialed number. [1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations
On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > hi, > > i want use asterisk+pjsip as voip client with multiple registrations > (perf testing) > > i'm using this example configuration for one account > > [308] > type=registration > outbound_auth=308 > server_uri=sip:3...@example.com:5060 > client_uri=sip:3...@example.com:5060 > > [308](auth-userpass) > username=308 > password=pass > > [308](aor-single-reg) > contact=sip:example.com:5060 > > [308](endpoint-basic) > outbound_auth=308 > aors=308 > > [308] > type=identify > endpoint=308 > match=example.com > > > my problem is contact on the other side (is same for all endpoints) > > Addr->IP : 1.1.1.1:5060 > Reg. Contact : sip:s@1.1.1.1:5060 > > all incoming calls from PBX to my Asterisk are routed to only one > account (because of same ip address/port ?) > > how can i specify different source port or different contact address for > asterisk pjsip client with registration? The "contact_user" option configures the user portion of the Contact that is sent in the REGISTER. You can set it to a different value for each registration. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk pjsip registration issues
On Tue, Sep 26, 2017, at 10:33 AM, Bryant Zimmerman wrote: > Hey all > > I am hoping someone can assist I have now spent over a week trying to > figure out what is going on with PJSIP registrations. > > I am able to register handsets against an asterisk 13 server running > pjsip, but I am not able to get pjsip to register out to an older > chan_sip > asterisk server. > If I drop the registration I can make things work, but when I have to > register the asterisk - pjsip server against another server the > registration completes, but I can not send any calls across the > registration, nor will it handle options correctly as well. > > We keep getting ... No auth credentials for realm(s) > 'aster...@xxx.xxx.xxx.xxx' in challenge. > in one form or another, and I have been unable to find any definitive > documentation on what is at cause for this. In some areas I have seen > responses saying it is an issue with realms so I have tried with and > without but no success. > > I really need some direction on this. This is the last issue I know of > that is holding up us from moving to pjsip. If I can't get asterisk / > pjsip > to register and send authenticated messages than it can't work for > replacing chan_sip in all situations. > > What am I doing wrong. Break this down further because you have some conflicting and confusing information. Does the outbound registration work or not work? Does it show as registered in PJSIP? If you leave out the "realm" option what happens? When you say "can't send any calls across the registration" what does that mean? Are you referring to inbound calls or outbound calls? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
On Fri, Sep 15, 2017, at 12:18 PM, Bryant Zimmerman wrote: > Joshua > > We are using MariaDB as the database storage. > We have recreated the database tables with alembic. > > Test 1: > We enable tables for aors, auths and endpoints only.With cache > turned > off the end point registers successfullyWe have no way to get any > feed > back as pjsip show/list returns no objects found. pjsip send notify > cmd > endpoint -- does not work as it says there is no endpoint. endpoint > can > send a call as it appears to be registered, we have no way to confirm > this > form the console but calls come in. The show and list commands are supposed to work, even without caching being enabled. Your problem is therefore at the realtime level. Calls coming in should appear on the console, and the endpoint name will be in the channel name. Enabling caching just masks it some because things exist in the cache for a bit. > > I can offer the following: > A dump of the database schema that alembic is creating. > extconfig.config > sorcery.conf Feel free to provide these and me (or another individual) may pick out what is wrong. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime pjsip issues
On Fri, Sep 15, 2017, at 11:38 AM, Bryant Zimmerman wrote: > Joshua > > We have completed more testing this morning and when we remove the > realtime cache options from the sorcery file the endpoints complete > registration, but we pjsip show/list does not offer any feed back at all, > We also can't send any pjsip send notify commands as they say they don't > have an endpoint there. Something has changed in the cache part of the > system that is breaking the system in some manner for us with the current > version and we are out of ideas. You're still confusing me here. If you've removed the cache, then it's not being used anymore so I don't see how it can be a problem. If the commands aren't listing things when you have no cache even in use then that would point to realtime, not the cache. You'd need to do as I said with debug to see what queries are being done to confirm things. You need to do troubleshooting and isolate things to determine the cause of the problem. You also did not answer my questions about the database schema. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
