Re: [asterisk-users] Adding Subscribe Handlers in PJSIP
Hi all, Is there a way with Polycom phones or alternatives, to configure a specific SIP server for such as-feature-event or call-info events ? If positive, maybe a third party SIP server (Kamailio, ...) supporting those events would allow such implementation. Looking at Yealink phone Admin guide, it seems possible to set a Yealink phone to handle such event Notify/Subscribe communicating with a dedicated SIP server. It seems rather "expensive" though. Thoughts ? Best regards Le mer. 1 mars 2017 à 21:21, Joshua Colp a écrit : > On Wed, Mar 1, 2017, at 04:02 PM, Trey Hilyard wrote: > > Is there any "easy" way to add a custom subscribe handler? I have a set > > of > > users with Polycom phones that attempt to Events that Asterisk/PJSIP > > doesn't recognize, "call-info" and "as-feature-event". It just generates > > a > > warning, but it got me wondering if I could add my own handlers for those > > that didn't actually do anything but simply responded with a 200 OK. > > > > Yes, I can probably stop the phones from subscribing, but this is more > > academic at this point. I assume there are things that I could do if I > > wanted to make changes and recompile, but is there an easier way to add a > > handler? I am a little confused about whether there is a subscribe > > handler > > in front of res_pjsip_pubsub, or if that is the first place that a > > SUBSCRIBE could get caught. > > The res_pjsip_pubsub module itself provides a framework for registering > support for event types (ast_sip_register_subscription_handler) and > handles the subscription lifetime. Callbacks are invoked on the various > things. Subscription requested, terminated, etc. There are also > functions for sending a NOTIFY and such. > > There's also another framework for handling PUBLISH. > > This all does require writing a C module though and building it with > Asterisk. There's no external mechanism to implement such 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 -- > > 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 > -- _ -- 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] Adding Subscribe Handlers in PJSIP
A site question: which of the following RFC would describe as-feature-event ? [1] https://www.iana.org/assignments/sip-events/sip-events.xhtml Le mer. 1 mars 2017 à 21:03, Trey Hilyard a écrit : > Is there any "easy" way to add a custom subscribe handler? I have a set of > users with Polycom phones that attempt to Events that Asterisk/PJSIP > doesn't recognize, "call-info" and "as-feature-event". It just generates a > warning, but it got me wondering if I could add my own handlers for those > that didn't actually do anything but simply responded with a 200 OK. > > Yes, I can probably stop the phones from subscribing, but this is more > academic at this point. I assume there are things that I could do if I > wanted to make changes and recompile, but is there an easier way to add a > handler? I am a little confused about whether there is a subscribe handler > in front of res_pjsip_pubsub, or if that is the first place that a > SUBSCRIBE could get caught. > > -Trey > -- > _ > -- 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 -- _ -- 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] Adding Subscribe Handlers in PJSIP
On Tue, Jan 15, 2019, at 9:29 AM, Olivier wrote: > A site question: which of the following RFC would describe as-feature-event ? > > [1] https://www.iana.org/assignments/sip-events/sip-events.xhtml If I recall correctly it doesn't have a spec, it's one of the custom things Broadsoft has done from their platform. -- Joshua C. 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 -- 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] MWI Delayed on Polycom VVX phones
Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in the change logs for 13.24.0 [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default Any suggestions on what to look at to diagnose? Doug -- _ -- 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] MWI Delayed on Polycom VVX phones
> Am 15.01.2019 um 15:23 schrieb Doug Lytle : > > Hi all, > > When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has > resulted in a MWI clearing delay of around 5 minutes. > > After listening to a voicemail and deleting it, the Polycom VVX 601's MWI > light is left on for around five minutes, before clearing. > > Installing Asterisk 13.24.1 did not fix this. > > Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in > the change logs for 13.24.0 > > [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead > of mailboxes=##@default > > Any suggestions on what to look at to diagnose? > > Doug Hi Doug, applying this patch helped in my case (with AstLinux 1.3.x + Asterisk 13.24.1): https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17 I am using "mailboxes=##@default" and had the issue as well (before). Michael http://www.mksolutions.info -- _ -- 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] MWI Delayed on Polycom VVX phones
>>> https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17 >>> I am using "mailboxes=##@default" and had the issue as well (before). >>> Michael Thanks Michael! I'll try that patch later on today. I'm not using the mailboxes=##, but will try the patch just the same. Doug -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1 date and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing for sure. NTPD is running, but not working for some reason. I fixed it (ownership of ntp.conf wrong) so now ntpq -pn returns a server ID. Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Stefan Viljoen Sent: Tuesday, January 15, 2019 12:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well. To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf: . . . load => res_timing_dahdi.so noload => res_timing_pthread.so noload => res_timing_timerfd.so That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines. We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had. So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything... Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it? Because the symptoms described could indicate that the clock in the VM is just running too fast - or that some timing implementation detail inside Asterisk itself is running too fast. Regards Stefan -- _ -- 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 -- _ -- 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
[asterisk-users] How to build and use your custom asterisk .deb package ?
Hello, There is question that bounces in my mind for quite a long time. Today, I dare to ask it here: how do you package and use your custom asterisk .deb package ? The background is: - I'm now a long time Debian user and I learned to appreciate Debian's deb package benefits specially when dealing with complex softwares such as Asterisk - On another hand, new Asterisk versions are regularly published. Looking at Debian's asterisk source packages, beside patching Asterisk source code, it seems possible to build a new Asterisk 13.X+1 package copying 13.X package - If possible (ie not too hard), I would be happy to build and maintain an Asterisk 16.X binary package for Buster (amd targets). So my questions are: 1. Would you evaluate porting Debian's patches from one Asterisk version to the next one (Asterisk 16.4 to 16.5, for example), to be a complex or time consuming task ? 2. What is the simplest and safest way to deal with the existence of both custom and original packages ? Using deb packages terminology, would you simply create a mycustom-asterisk package which both conflicts and provides asterisk, if that is possible ? Alternatively, is apt-pinning recommended ? 3. Can I realistically hope, that I won't have to change Asterisk 16 dependencies during Asterisk evolution, at least for Asterisk core features or I shall prepare to also upgrade some libraries ? 4. Suggestions ? Pointers ? Best regards -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
>>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
Actually, I was wrong about that. We no longer use OVM. It's actually Citrix Xencenter 7.6 Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: Tuesday, January 15, 2019 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters >>> Carlos and Stefan (and other who have helped): Thomas, You stated that your virtual environment was Oracle, would that equate to VirtualBox? Doug -- _ -- 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 -- _ -- 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
[asterisk-users] what service does asterisk need to avoid netsock error ?
I'm running Fedora 29. asterisk starts with a systemd service at boot. On any reboot I get a LOT of : [Jan 15 09:30:26] ERROR[1162]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. [Jan 15 09:30:35] ERROR[1161]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. [Jan 15 09:30:45] ERROR[1155]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. [Jan 15 09:30:45] ERROR[1155]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'. If I "systemctl restart asterisk", no more errors. I think this means something needs to start before asterisk starts. My asterisk.service has: [Unit] Description=Asterisk PBX and telephony daemon. After=network.target but obviously that's not enough. What else needs to start before asterisk ? -- _ -- 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] Various extensions ring once and go to voicemail - Thomas Peters
From https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces: res_timing_dahdi uses timing mechanisms provided by DAHDI. This method of timing was previously the only means by which Asterisk could receive timing. It has the benefit of being efficient, and if a system is already going to use DAHDI hardware, then it makes good sense to use this timing source. If, however, there is no need for DAHDI other than as a timing source, this timing source may seem unattractive. For users who are upgrading from Asterisk 1.4 and are used to the ztdummy timing interface, res_timing_dahdi provides the interface to DAHDI via the dahdi kernel module. res_timing_timerfd uses a timing mechanism provided directly by the Linux kernel. This timing interface is only available on Linux systems using a kernel version at least 2.6.25 and a glibc version at least 2.8. This interface has the benefit of being very efficient, but at the time this is being written, it is a relatively new feature on Linux, meaning that its availability is not widespread. On 01/15/2019 09:53 AM, Thomas Peters wrote: Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0. NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so. Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind? SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did watch -n1 date and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing for sure. NTPD is running, but not working for some reason. I fixed it (ownership of ntp.conf wrong) so now ntpq -pn returns a server ID. Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Milwaukee County Transit System 1942 N 17th Street | Milwaukee, WI 53205 Check us out on Facebook & Twitter -Original Message- From: asterisk-users On Behalf Of Stefan Viljoen Sent: Tuesday, January 15, 2019 12:05 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules loaded So what would you suggest? (And thanks in advance.) Thomas I've had some good experience with res_timing_dahdi both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well. To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf: . . . load => res_timing_dahdi.so noload => res_timing_pthread.so noload => res_timing_timerfd.so That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines. We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had. So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything... Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it? Because the symptoms described could indicate that the clock in the VM is just running too fast - or that some timing implementation detail inside Aste
[asterisk-users] (NAT) direct media to host on local net when registering from external address
This is going to be a bit of an odd situation, but perhaps might become more and more common (as mobile phone SIP clients utilize PUSH proxies instead of the battery draining direct registering with SIP servers). I have a SIP client which can be on the same RFC-1918 LAN as my Asterisk server. Even though it's on the same LAN as the Asterisk server, it's registration comes from an IP address external to the LAN. This is because the client is registering to the local Asterisk server through a SIP proxy server that is external to the LAN. Is there any way for Asterisk to determine that this is what is happening and to direct/setup the media session to the client on it's LAN address? Put another way, even though the registration comes from an external (NATted) IP address, I want the media connection to stay within the LAN. One solution of course is to add the external IP address of the SIP proxy -- the address that the client's registration is coming from -- to localnet but that breaks the use-case of the SIP client (which is mobile) leaving the LAN and having an external IP address. Ultimately I am hoping there is something in the registration that will indicate to Asterisk that even though the registration is coming from an external IP address, the client has an internal IP address that is on the same network as it is and sets the IP address in the SDP payload to a local IP address. I do realize this is fairly non-standard configuration so it might not be possible. Any suggestions would be welcome. Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- 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] (NAT) direct media to host on local net when registering from external address
How is your endpoint currently configured in asterisk? Have you tried rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk can send audio back to the client? Alternatively if your SIP Proxy is also a Media proxy you could set the media_address on the endpoint to be your proxy and let your proxy handle proxying the media to your endpoint. On Tue, Jan 15, 2019 at 8:47 AM Brian J. Murrell wrote: > > Put another way, even though the registration comes from an external > (NATted) IP address, I want the media connection to stay within the > LAN. > -- _ -- 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] (NAT) direct media to host on local net when registering from external address
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > How is your endpoint currently configured in asterisk? It's configured as a chan_sip peer. > Have you tried > rtp_symmetric to see if the endpoint sends audio to asterisk if > asterisk > can send audio back to the client? That would require using chan_pjsip wouldn't it? Not that I am opposed to trying that. I need to use chan_pjsip at some point to be able to authenticate to my SIP provider for SIP SIMPLE anyway. But will rtp_symmetric really solve the problem? Isn't the problem the setting up of the RTP session, so there is no address and port that it receives from yet? > Alternatively if your SIP Proxy is also a Media proxy you could set > the > media_address on the endpoint to be your proxy and let your proxy > handle > proxying the media to your endpoint. The idea of sending my media out of the LAN (where I have almost zero latency) and introducing the latency of a round trip to the proxy and back doesn't seem like a great solution. Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- 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
[asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly?
Hi Guys
I've run into a weird problem on Asterisk 13. Again something that worked fine
on 1.8 but is now broken on Asterisk 13.
I have an extension 3015. I'm trying to originate a recording playback call on
it via AMI by sending
Action: Originate
ActionID: test
Channel: SIP/3015
Exten:
Context: local
Priority: 1
CallerID: 3015
Account: recordinglisten
ChannelID: abc
OtherChannelID: def
Variable:
CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
Async: true
My dialplan code:
exten=>,1,Answer()
exten=>,n,NoOp(Requesting File ${recfile})
exten=>,n,Set(${__recfile}=${recfile})
exten=>,n,Set(${__altfile}=${altfile})
exten=>,n,NoOp(Rec file set to ${recfile})
exten=>,n,NoOp(Alt file set to ${altfile})
exten=>,n,NoOp(Requesting Alt File ${altfile})
exten=>,n,Set(__numbertarget=)
exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
exten=>,n(play2),ControlPlayback(${altfile},2,6,4,8,5,9)
exten=>,n,hangup()
However, on sending the above to the AMI I get in the console of Asterisk 13:
[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
local
and no recording playback takes place.
I've found that by editing /etc/hosts and adding
172.56.4.11 local
where 172.56.4.11 is the phone SIP/3015's IP address, it works perfectly and
the recording starts playing once 3015 is answered.
On Asterisk 1.8 the above worked without having to constantly edit /etc/hosts
and having to constantly map and remap the one phone that is then capable of
playing back a recording.
This appears to be some kind of DNS / name resolution issue exclusive to
Asterisk 13 (for me) - how can I fix this, e. g. NOT get
[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
local
and constantly having to edit /etc/hosts to get one of my phones (the one tied
to "local") to be able to play back a recording on Asterisk 13?
(This obviously is fatal anyway as I got lots of phones on which I want to
playback recordings and editing /etc/hosts for each phone is impossible if two
phones want to listen to different recordings at the same time- /etc/hosts can
only contain one "local").
How can I fix this?
Thanks
Stefan
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Re: [asterisk-users] Various extensions ring once and go to voicemail
Subject: Re: [asterisk-users] Various extensions ring once and goto voicemail - Thomas Peters >Carlos and Stefan (and other who have helped): >I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling >Asterisk is unrealistic in my position but I wonder if I can build the one >module. Here's what I do have: >apbx:~ $ locate *res_timing_timerfd* >/usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts >/usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo >/usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c >/usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts >/usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo >/usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c >Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version >1.8.7.0. Ouch. Sounds like you're maybe sitting with a hybrid package install setup and a partial source based install - did you setup this box yourself? It appears that the binaries of asterisk were compiled, then the source was deleted or the binaries that comprise your instance were installed from a package... You can probably build only timerfd, but it does imply running menuconfig (I think) and for that you need a properly configured Asterisk source tree, of the correct version you want. >NEXT QUESTION: There are NO timing modules listed in >/etc/asterisk/modules.conf at all. The only ones that are explicitly loaded >are format_wav format_pcm format_mp3 and res_musiconhold. And there are >"preload" directives for pbx_config.so and chan_local.so. >Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of >some kind? AFAIK it is a default, but as default, again AFAIK, res_timing_dahdi.so won't get loaded, the pthread timer or timerfd will be used. Since you don't even have the timerfd module, you are running by implication on timerfd. >SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did >watch -n1 date >and watched the time tick up, perfectly synchronized to my mobile phone. It >might be off by a second or so, I'd have a hard time knowing for sure. NTPD is >running, but not working for some reason. I fixed it (ownership of ntp.conf >wrong) so now ntpq -pn returns a server ID. Ok... well scratch that theory then. As I said earlier, we have had very strange misbehaviour with Asterisk in virtually hosted environments, and after bitter experience resolved to run it only in real physical boxes as it seems to perform best there and be the most stable and reliable. All I might suggest is getting the latest asterisk source in the 1.8 series (1.8.32.2 if I'm not mistaken - we ran it for years) and compile it from scratch. But do not install it, e. g. if you do make install it will overwrite your current setup irretrivably. Rather, compile it, and then make actual physical copies of your current asterisk binary (/usr/sbin/asterisk, I think) and of your /usr/lib/asterisk/modules folder, -then- make install it. Start it up and see if it works better. If it is a success, great. If not, simply copy back your copied asterisk binary and copy back all the files in /usr/lib/asterisk/modules, and restart your old version which at least is working partially. Again, no guarantees, the fact that you apparently already have a disjointed setup (at least three asterisk versions?) might mitigate against this - your milage may vary and doing what I describe might also destroy your entire current setup. The reader must beware. It sounds as if you will need to recompile Asterisk from a known clean source to begin troubleshooting it anyway? Kind regards, Stefan -- _ -- 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] (NAT) direct media to host on local net when registering from external address
On Tue, Jan 15, 2019, at 12:18 PM, Brian J. Murrell wrote: > On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote: > > How is your endpoint currently configured in asterisk? > > It's configured as a chan_sip peer. > > > Have you tried > > rtp_symmetric to see if the endpoint sends audio to asterisk if > > asterisk > > can send audio back to the client? > > That would require using chan_pjsip wouldn't it? Not that I am opposed > to trying that. I need to use chan_pjsip at some point to be able to > authenticate to my SIP provider for SIP SIMPLE anyway. > > But will rtp_symmetric really solve the problem? Isn't the problem the > setting up of the RTP session, so there is no address and port that it > receives from yet? The chan_sip module has this implemented under the "nat" option using "comedia" as I recall. It causes media to be sent to where media was originally received from. As for whether it would work or not... this all ultimately depends on how exactly the intermediary behaves, what is let through, what is altered. There's nothing inherent in either chan_sip or chan_pjsip to know/care, as it's just SIP. You'd need to look at the SIP signaling and the SDP to understand what is happening. -- Joshua C. 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 -- 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] MWI Delayed on Polycom VVX phones
>>> I'll try that patch later on today. I'm not using the mailboxes=##, but >>> will try the patch just the same. Patch applied and fixed my problem, Doug -- _ -- 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
[asterisk-users] Cross-compiling Asterisk 16
Hello, I've just gone through the process of cross-compiling Asterisk 16 for ARM. I thought it would be as easy as calling the "./configure" script with the appropriate "host" parameter, but it turned out to be more complicated. I'm wondering whether I did something wrong, or if there are some bugs in the "configure" script. Here are the issues I had to solve : 1- in the "configure" script, checking the presence of "hrirs.h" does not work when cross-compiling. I had to comment out all the related code in the script. 2- pkg-config does not always use the proper directories. "Not always" because it worked properly on Ubuntu, but not on Centos. I had to manually set PKG_CONFIG_LIBDIR and PKG_CONFIG_PATH appropriately when launching "configure" 3- pjproject and jansson are not properly cross-compiled. It looks like the target architecture is not properly set during the configuration process. I had to reconfigure them manually before compiling. Does anybody have any experience with this ? And should I fill bug reports for these issues ? Regards Jean Aunis -- _ -- 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] (NAT) direct media to host on local net when registering from external address
On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote: > > The chan_sip module has this implemented under the "nat" option using > "comedia" as I recall. Yeah. The help for which reads: Send media to the port Asterisk received it from regardless of where the SDP says to send it. > It causes media to be sent to where media was originally received > from. Right. But this is the part I don't think I'm understanding. What is "originally received from"? There is no media at the point where the media session is being set up according the the SDP is there? > As for whether it would work or not... this all ultimately depends on > how exactly the intermediary behaves, what is let through, what is > altered. There's nothing inherent in either chan_sip or chan_pjsip > to know/care, as it's just SIP. You'd need to look at the SIP > signaling and the SDP to understand what is happening. I have looked. Asterisk on my LAN is sending an INVITE to the mobile client and since the INVITE is being sent to an external IP, it is (correctly for all other cases) writing the external IP into the SDP payload. What else could/should I be looking for? Ultimately I don't think comedia is the problem or solution because it's not where Asterisk is sending it's media that is the problem. The problem is that Asterisk is sending an INVITE to the client with it's external IP address in the SDP because it sees the client as being external and the client is then correctly trying to set up the media session with the external IP address of the Asterisk instead of it's internal address. What needs to happen is that Asterisk has to know "somehow" that even though the client registered from an external address, that it really is internal. Probably that "somehow" doesn't actually exist at this point. Let me ask this, in the REGISTER request, where is the IP address of the client taken from? Is it taken from the Via: header? b. signature.asc Description: This is a digitally signed message part -- _ -- 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] (NAT) direct media to host on local net when registering from external address
On Tue, Jan 15, 2019, at 1:17 PM, Brian J. Murrell wrote: > On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote: > > > > The chan_sip module has this implemented under the "nat" option using > > "comedia" as I recall. > > Yeah. The help for which reads: > > Send media to the port Asterisk received it from regardless > of where the SDP says to send it. > > > It causes media to be sent to where media was originally received > > from. > > Right. But this is the part I don't think I'm understanding. What is > "originally received from"? There is no media at the point where the > media session is being set up according the the SDP is there? It waits until the RTP flows to determine this. > > > As for whether it would work or not... this all ultimately depends on > > how exactly the intermediary behaves, what is let through, what is > > altered. There's nothing inherent in either chan_sip or chan_pjsip > > to know/care, as it's just SIP. You'd need to look at the SIP > > signaling and the SDP to understand what is happening. > > I have looked. Asterisk on my LAN is sending an INVITE to the mobile > client and since the INVITE is being sent to an external IP, it is > (correctly for all other cases) writing the external IP into the SDP > payload. What else could/should I be looking for? > > Ultimately I don't think comedia is the problem or solution because > it's not where Asterisk is sending it's media that is the problem. > > The problem is that Asterisk is sending an INVITE to the client with > it's external IP address in the SDP because it sees the client as being > external and the client is then correctly trying to set up the media > session with the external IP address of the Asterisk instead of it's > internal address. Ah yes... yeah, there's nothing built in to handle the scenario. > What needs to happen is that Asterisk has to know "somehow" that even > though the client registered from an external address, that it really > is internal. Probably that "somehow" doesn't actually exist at this > point. > > Let me ask this, in the REGISTER request, where is the IP address of > the client taken from? Is it taken from the Via: header? For what purpose? The SIP response goes to the source of the request. If you mean the registration address, that is the Contact header. A REGISTER essentially means "I can be reached at this Contact address for requests". -- Joshua C. 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 -- 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 originate to extension unless /etc/hosts is edited constantly?
In article <018201d4acef$898a4b10$9c9ee130$@verishare.co.za>,
Stefan Viljoen wrote:
> Hi Guys
>
> I've run into a weird problem on Asterisk 13. Again something that worked
> fine on 1.8 but is now broken on Asterisk 13.
>
> I have an extension 3015. I'm trying to originate a recording playback call
> on it via AMI by sending
>
> Action: Originate
> ActionID: test
> Channel: SIP/3015
> Exten:
> Context: local
> Priority: 1
> CallerID: 3015
> Account: recordinglisten
> ChannelID: abc
> OtherChannelID: def
> Variable:
> CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
> Async: true
>
> My dialplan code:
>
> exten=>,1,Answer()
> exten=>,n,NoOp(Requesting File ${recfile})
> exten=>,n,Set(${__recfile}=${recfile})
> exten=>,n,Set(${__altfile}=${altfile})
> exten=>,n,NoOp(Rec file set to ${recfile})
> exten=>,n,NoOp(Alt file set to ${altfile})
> exten=>,n,NoOp(Requesting Alt File ${altfile})
> exten=>,n,Set(__numbertarget=)
> exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
> exten=>,n(play2),ControlPlayback(${altfile},2,6,4,8,5,9)
> exten=>,n,hangup()
>
> However, on sending the above to the AMI I get in the console of Asterisk 13:
>
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
> local
>
> and no recording playback takes place.
>
> I've found that by editing /etc/hosts and adding
>
> 172.56.4.11 local
>
> where 172.56.4.11 is the phone SIP/3015's IP address, it works perfectly and
> the recording starts playing once 3015 is answered.
What does your sip.conf look like? (without comments, e.g. grep -v '^;'
sip.conf | grep -v '^$')
Particularly, do you have a separate section for each phone, e.g. [3015] ?
> On Asterisk 1.8 the above worked without having to constantly edit /etc/hosts
> and having to constantly map and remap the one phone that is then capable of
> playing back a recording.
>
> This appears to be some kind of DNS / name resolution issue exclusive to
> Asterisk 13 (for me) - how can I fix this, e. g. NOT get
>
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
> local
>
> and constantly having to edit /etc/hosts to get one of my phones (the one
> tied to "local") to be able to play back a recording on Asterisk 13?
>
> (This obviously is fatal anyway as I got lots of phones on which I want to
> playback recordings and editing /etc/hosts for each phone is impossible if
> two phones want to listen to different recordings at the same time-
> /etc/hosts can only contain one "local").
>
> How can I fix this?
I don't understand why it should be trying to look up the address of "local".
If you change your dialplan context to another name and refer to that name in
Context in your original request, does it try looking up this new name, or
still "local"?
Cheers
Tony
--
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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Re: [asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly? [Tony Mountfield]
Hi Tony
Ok, got this solved.
I discovered my AMI message was corrupt due to a bug in our third party dialer
app we wrote ourselves...!
E. g. this worked on Asterisk 1.8:
ActionID=12edad43-e817-427b-aa21-31a9659f86e1
&Action=Originate
&Channel=SIP/local/3035@local
&Exten=
&Context=local
&Priority=1
&CallerID=3035
&Account=recordinglisten
&ChannelID=12edad43-e817-427b-aa21-31a9659f86e1
&OtherChannelID=12edad43-e817-427b-aa21-31a9659f86e1B
&Variable=CallLimit=3600,recfile=/var/spool/asterisk/monitor/archive/1901/15/201901151654r1g4679,altfile=/var/spool/asterisk/monitor/archive/1901/15/201901151654r1g4679
&Async=true
But doesn't on Asterisk 13 - for a very good reason:
&Channel=SIP/local/3035@local
was acceptable to Asterisk 1.8, but NOT to Asterisk 13 - and I kind of agree.
It was a bug in our third party dialer app that instead of passing to Asterisk
13
&Channel=SIP/3035@local
was passing
&Channel=SIP/local/3035@local
which is just WRONG.
This was why 13 was replying
[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
local
which makes perfect sense...!
So I fixed the incorrect variable reference in our app that was generating the
wrong
SIP/local/3035@local
channel, and recording playback started working correctly and no more of the
above error messages.
So all not Asterisk fault, though 13 does appear to interpret the channel name
differently as
SIP/local/3035@local
DOES work on 1.8.32.3 (which it shouldn't, but it does.)
Thanks for taking the time to reply.
Regards
Stefan
---
Date: Tue, 15 Jan 2019 17:32:40 + (UTC)
From: t...@softins.co.uk (Tony Mountifield)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot originate to extension unless
/etc/hosts is edited constantly?
Message-ID:
In article <018201d4acef$898a4b10$9c9ee130$@verishare.co.za>,
Stefan Viljoen wrote:
> Hi Guys
>
> I've run into a weird problem on Asterisk 13. Again something that worked
> fine on 1.8 but is now broken on Asterisk 13.
>
> I have an extension 3015. I'm trying to originate a recording playback call
> on it via AMI by sending
>
> Action: Originate
> ActionID: test
> Channel: SIP/3015
> Exten:
> Context: local
> Priority: 1
> CallerID: 3015
> Account: recordinglisten
> ChannelID: abc
> OtherChannelID: def
> Variable:
> CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
> Async: true
>
> My dialplan code:
>
> exten=>,1,Answer()
> exten=>,n,NoOp(Requesting File ${recfile})
> exten=>,n,Set(${__recfile}=${recfile})
> exten=>,n,Set(${__altfile}=${altfile})
> exten=>,n,NoOp(Rec file set to ${recfile})
> exten=>,n,NoOp(Alt file set to ${altfile})
> exten=>,n,NoOp(Requesting Alt File ${altfile})
> exten=>,n,Set(__numbertarget=)
> exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
> exten=>,n(play2),ControlPlayback(${altfile},2,6,4,8,5,9)
> exten=>,n,hangup()
>
> However, on sending the above to the AMI I get in the console of Asterisk 13:
>
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
> local
>
> and no recording playback takes place.
>
> I've found that by editing /etc/hosts and adding
>
> 172.56.4.11 local
>
> where 172.56.4.11 is the phone SIP/3015's IP address, it works perfectly and
> the recording starts playing once 3015 is answered.
What does your sip.conf look like? (without comments, e.g. grep -v '^;'
sip.conf | grep -v '^$')
Particularly, do you have a separate section for each phone, e.g. [3015] ?
> On Asterisk 1.8 the above worked without having to constantly edit /etc/hosts
> and having to constantly map and remap the one phone that is then capable of
> playing back a recording.
>
> This appears to be some kind of DNS / name resolution issue exclusive to
> Asterisk 13 (for me) - how can I fix this, e. g. NOT get
>
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve:
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host:
> local
>
> and constantly having to edit /etc/hosts to get one of my phones (the one
> tied to "local") to be able to play back a recording on Asterisk 13?
>
> (This obviously is fatal anyway as I got lots of phones on which I want to
> playback recordings and editing /etc/hosts for each phone is impossible if
> two phones want to listen to different recordings at the same time-
> /etc/hosts can only contain one "local").
>
> How can I fix this?
I don't understand why it should be trying to look up the ad
