Re: [asterisk-users] Questions about sRTP

onal *SRTP*. If a device requests SRTP and it's not possible, the call will fail. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com

Re: [asterisk-users] Google/XMPP and Asterisk/XMPP

ejabberd The XMPP support is not tuned for Google Talk by any means, and the voice part (chan_motif) supports the two Google derivatives and the actual Jingle spec. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at

Re: [asterisk-users] asterisk debian package and digium repository

ge in the situation. The asterisk.org Debian packages have not been updated. The debian.org packages are done by others, so I can't speak on that. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.

Re: [asterisk-users] Question

erminated your access. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provide

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

sean darcy wrote: I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? Motif itself has no imposed limitations, but that's not to say Google Voice doesn't.

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

Matthew J. Roth wrote: Joshua Colp wrote: If you set nat=no for that specific peer it should work as you need. 'rport' is forced on these days which works for most situations, except with some platforms and Cisco phones.>_> Joshua, That sounds much easier than what I came

Re: [asterisk-users] On SIP INVITE answering to IP:port found in Contact: header.

rced on these days which works for most situations, except with some platforms and Cisco phones. >_> -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:

Re: [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee

l) Call-ID: 705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060 <http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060/>. The call id was changed twice Could this be a two part problem? Yes. Until you can isolate it more it's all just a guess but it still doesn't seem lik

Re: [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee

races also illustrate this, the BYE in the trace is from a completely different call than the other messages. (You can see by looking at the Call-ID). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digiu

Re: [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee

user agent. Each leg is individual. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

Daniel Pocock wrote: On 01/04/13 22:06, Joshua Colp wrote: Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

VP or SAVP." and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

owledge of AVPF, and since it's a new feature it was only added to Asterisk 11. You could try to backport the changes but chan_sip has changed quite a bit, so it could be rough. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check u

Re: [asterisk-users] Optimizing Asterisk Environment

d the only option to make it go further is to use a hardware transcoding solution. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com

Re: [asterisk-users] Asterisk 11 & GoogleVoice/Motif

f Asterisk seems to clear it up for another day or so. Has anyone else noticed this? This is fixed in Asterisk 11.3.0-rc1. The Google XMPP server has become prone to disconnecting as of late, which triggers a bug in older versions where chan_motif ignores XMPP traffic. -- Joshua Colp D

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

=> n(email),System(/usr/local/bin/emailme) same => n,Answer() ; also tried without this same => n,Hangup() You need to Answer, Wait, send a DTMF of 1, wait a bit more, and then hang up. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW -

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

7;t specified anything to call. You need to specify both an endpoint and a target to call. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digiu

Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif

f the Google Talk XMPP server. No incoming connections occur. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ --

Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif

s nothing that can be done to force them to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandw

Re: [asterisk-users] Google voice with no voice

/display/AST/Calling+using+Google -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Coloca

Re: [asterisk-users] Google voice with no voice

and loaded. You can load it explicitly using "module load app_senddtmf.so". If that fails then it was not built and you will have to look into why not. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Google voice with no voice

terisk dialplan with a combination of Answer, Wait, and SendDTMF(1) -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org --

Re: [asterisk-users] gtalk only working with ulaw???

Daniel Pocock wrote: On 14/01/13 23:31, Joshua Colp wrote: Daniel Pocock wrote: I've set up a peer to use G.722 only and tried to make it talk to an Asterisk box Asterisk always rejects the call with the following error: chan_gtalk was written to only support a limited number of codecs

Re: [asterisk-users] gtalk only working with ulaw???

does not have this limitation. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

Roy Abshire wrote: What is the quickest way to apply this patch? The quickest way is to probably just grab res_xmpp.c from http://svn.digium.com/svn/asterisk/branches/11/res/res_xmpp.c - replace the existing res_xmpp.c in your code with it - recompile - and install. -- Joshua Colp Digium

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

?r1=378411&r2=378917&view=patch and apply it against your Asterisk 11 source code. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

Roy Abshire wrote: Could this be why incoming calls to voice do not ring asterisk extensions too? When the issue occurs, yes. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

Joshua Colp wrote: Kai-Uwe Jensen wrote: On Wed, Jan 9, 2013 at 5:38 PM, Roy Abshire mailto:r...@coopvr.com>> wrote: I have the transport=google-v1 too but restarting Asterisk always solves my problem for a day...so how do you know that fixed it? I don't. If any of you can

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

ebug like I previously mentioned then I can see pretty quickly where the problem lies (chan_motif or Google Voice). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & w

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

curs hang up and run "xmpp set debug off". Copy the contents of the console to a file and put it somewhere for viewing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

k fixes it only for a day. Without seeing an XMPP debug with the Jingle signaling it's impossible to really investigate this. I will say I haven't heard of this happening though from anyone else. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsv

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

in the configuration. I would suggest you follow https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support since there may be other things you have missed. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] call recording via 3rd INVITE/SIP leg

ence bridge you can use ChanSpy to spy on a channel. It will provide a mixed stream of the incoming and outgoing part of the channel. So essentially use Originate to call your 3rd leg and then have it execute ChanSpy with the correct criteria to get to the right leg. Cheers, -- Joshua Colp D

Re: [asterisk-users] Queue joinempty, even after AddQueueMember

Jonas Kellens wrote: On 09-12-12 19:49, Joshua Colp wrote: As well - if the log you provided has not been altered then you are attempting to add an interface "member3" to the queue. While this will succeed it is ultimately not a valid interface and would not be considered as avail

Re: [asterisk-users] Queue joinempty, even after AddQueueMember

stable, it's just a candidate for release put out there for testing. It helps to uncover issues. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org --

Re: [asterisk-users] google talk under asterisk 11.0.1

provide some further insight. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Coloc

Re: [asterisk-users] app.c: No audio available on SIP

. On a functioning system audio is generally received every 20ms, or higher if packetization permits it. This could be a NAT problem or something else. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com

Re: [asterisk-users] SIP not answering on one trunk.

o from the log you have provided thus far, I think you may have a phone specific issue. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & ww

Re: [asterisk-users] SDK for Asterisk, where is it?

? The only thing that is remotely coming to mind is the SDK for the Digium Phones. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.ast

Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone

How is the call from your PSTN phone being delivered into Asterisk? If coming in over SIP the configuration may be incorrect. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.

Re: [asterisk-users] Google Voice Routing

Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 26/11/2012 04:26, Joshua Colp a écrit : To others using chan_motif - are you experiencing the same issue? I didn't use chan_motif since testing a few weeks ago, so I may I have broke my configuration, but Google

Re: [asterisk-users] Wierd RTP issue

little bit and caused the problem you heard (or didn't hear, hehe). If you still want to allow GSM you can try moving the allow=gsm to below allow=ulaw. This should change the priority. Glad it seems to be working for you now, though! Cheers, -- Joshua Colp Digium, Inc. | Senior Sof

Re: [asterisk-users] Wierd RTP issue

th phones and the desktop were in the same room). Few suggestions: 1. Remove allow=gsm from your sip.conf and reload 2. Disable ZRTP in Jitsi by going into Options -> Accounts -> Selecting account -> Edit -> Security -> Uncheck "Enable support to encrypt calls". S

Re: [asterisk-users] Wierd RTP issue

rk topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Hunts

Re: [asterisk-users] Wierd RTP issue

;SIP or DAHDI->SIP (and directmedia is off). Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA C

Re: [asterisk-users] Wierd RTP issue

ulaw allow=g729 allow=h264 What NAT settings are globally in use? Do you have directmedia turned off or on? This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). -- Joshua Colp Digium, Inc. | Senior Software Develope

Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

isr...@gmail.com wrote: Hi, If were on this subject I'll throw in my question Does named acl lists in asterisk 11 help for this or only for registrations? ACLs don't control SIP peer matching, so no. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW -

Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

d not recommend that as it can pose a security risk. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.ast

Re: [asterisk-users] Google Voice Routing

e sure the session initiation is proper. I'll see if I can reproduce this over the next few days in my spare time. To others using chan_motif - are you experiencing the same issue? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] Google Voice Routing

Chris Datfung wrote: On Mon, Nov 26, 2012 at 2:19 PM, Joshua Colp mailto:jc...@digium.com>> wrote: I'm trying to get Incoming Google Voice calls to ring on my Iaxy. I'm using Asterisk 11.0.1. Based on the the following configurations can someon

Re: [asterisk-users] Incorrect DTMF detection in Asterisk 1.8

ed at any time. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provide

Re: [asterisk-users] Meetme on short network

and have it pass through a jitterbuffer (which by definition of being a buffer introduces delay). How much of a delay are you hearing? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Wierd RTP issue

s silent when it's called from something on the other side of a PRI via DAHDI. What's the configuration like for Jitsi in sip.conf? What version of Asterisk? What does the SIP signaling look like? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW

Re: [asterisk-users] Google Voice Routing

fully load? If it didn't it would not attach itself to your Google account, so incoming session creation attempts would be ignored. Are there additional parts to your configuration files? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville,

Re: [asterisk-users] core show translation - difference in Asterisk Versions

e now seeing above is actually the internal "table cost" for choosing. I can certainly agree that this is not ideal. The subject has been brought up a few times but nobody has tackled making it reflect computational costs once again. Cheers, -- Joshua Colp Digium, Inc. | Senior Softwar

Re: [asterisk-users] watchdog like functions

"IAX trunk drops"? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Coloc

Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

ar to have been smooth. Would you be willing to provide the information I asked about from a running 11 instance? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com &am

Re: [asterisk-users] No more connections allowed.

houldn't happen but it would be useful to know exactly what you are doing with the system. Answering my questions above is a good start. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com

Re: [asterisk-users] addressing peers dynamically

devices. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

ody has responded to you until now. If you can provide that information I'm sure we can all help to determine if there really is an issue at work here! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Noise on phones while speaking...

post a short snippet of a phone calling another and the bridge that occurs I could be more certain. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] meetme race condition

nce bridge first. This is what Page essentially does, with the difference being that only the channel executing Page() can talk. If that behavior is what you are trying to accomplish I suggest you use that instead. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Dr

Re: [asterisk-users] addressing peers dynamically

peers are registering to that queue. is that the right path, or am i barking the wrong tree? Is there any particular reason you don't want to do this or is it just because you get the "Unable to create channel" message? There's nothing really *wrong* with that message in y

Re: [asterisk-users] Allowing peers from specific subnet only

172.16.10.1-172.16.10.255 range. For cases where you may want to configure this in one place and share it around Asterisk 11 has introduced what are called "Named ACLs". You can find further information on those at https://wiki.asterisk.org/wiki/display/AST/Named+ACLs Cheers, -- J

Re: [asterisk-users] Web based Click to Call Application

ng way but are still in a great state of flux. It's early for WebRTC. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com &am

Re: [asterisk-users] Managing complex setups with Asterisk

martin f krafft wrote: also sprach Joshua Colp [2012.11.07.1831 +0100]: Peer names have to be distinct, this is just a fundamental design element of chan_sip. What a lot of people end up doing is instead of treating peers as people they treat them as devices. The peer name becomes the MAC

Re: [asterisk-users] Managing complex setups with Asterisk

er at sip.conf.sample - specifically the SIP DOMAIN SUPPORT section and you'll see what I mean. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & ww

Re: [asterisk-users] Can you help me to use SIPML5 with Asterisk ?

dentity: Public Identity: sip:@of Asterisk> Password: Realm: In the future please send emails of this type to the asterisk-users mailing list so that everyone can see the conversation and learn. I've copied my reply to it. Cheers, -- Joshua Colp Digium, Inc. | Senior Sof

Re: [asterisk-users] 503 unable to load

an issue within Asterisk. This is a problem with the SIP destination you are dialing. Something is not configured properly with it or it is experiencing issues. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:

Re: [asterisk-users] Google Voice and back (chan_motif)

t, and Google Talk/Google Voice don't use ICE-UDP candidates. Sorry for the inconvenience! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- ___

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

problem knowing what Google Talk Login to use? There's nothing explicit to prevent you from doing this but Google decides what client gets incoming calls, so it may go to your desktop when you don't want it to. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

ng stands out immediately. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

dd icesupport=yes. I use my own rtp port range that is opened on the firewall. Yes, this is indeed a requirement. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www

Re: [asterisk-users] dialplan reloading

ported branches (1.8, 10, and trunk). http://lists.digium.com/pipermail/asterisk-commits/2012-February/053537.html For the 1.8 fix if you are curious. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:

Re: [asterisk-users] SIP - Authenticated vs Unauthenticated Calls

se Asterisk would simply throw it into the from-sip-external context as an unknown/unauthenticated call? And of course, when the peer *is* registered, and a call is made, Asterisk on the central system allows the call as authenticated due to the source IP/port being known via the registration status

Re: [asterisk-users] SIP - Authenticated vs Unauthenticated Calls

carries caller ID number information, with can obviously change between calls. That's my best guess without "sip set debug on" output for a non-working call and the configuration. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Hu

Re: [asterisk-users] DTMF Payload Settings

Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www

Re: [asterisk-users] Asterisk does not re-register as a sip client after a "sip reload" if "sip.conf" or "users.conf" is changed

://issues.asterisk.org/jira/browse/ASTERISK-20611 Is there a good reason you'd release 11.0 today with this serious a bug still in it? Asterisk 11 was made available Thursday of last week before this was known, the announcement was delayed due to AstriCon. -- Joshua Colp Digium, Inc. | S

Re: [asterisk-users] Asterisk does not re-register as a sip client after a "sip reload" if "sip.conf" or "users.conf" is changed

rogress on solving it at https://issues.asterisk.org/jira/browse/ASTERISK-20611 Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _

Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?

else they are mismatched and like you have seen, the universe will explode. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.d

Re: [asterisk-users] Add a variable to the destination channel without adding it to the source channel?

t people want to do varies greatly. I hope this helps! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _

Re: [asterisk-users] motif and psi - no sound

you've created an issue for this, thanks! I took a quick gander at the packet capture and it appears that the Jingle and RTP portion of things is perfectly fine. This may actually end up being a transcoding issue, but we'll see once the issue is assigned and researched. Cheers,

Re: [asterisk-users] DTMF inband with telephone-event in SDP

. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.a

Re: [asterisk-users] RC2, was: Motif/XMPP for Google Voice

ogle Voice would not work due to ignoring candidates. Instead of ignoring parts of the message that are not known just ignore the ones we know may be present and that would cause a problem. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davi

Re: [asterisk-users] Motif/XMPP for Google Voice

io working - issues were discovered that have been fixed. These fixes will be in the next release candidate. We also changed a default setting to no for ICE support, this needs to be set to yes for chan_motif to operate and I have updated the wiki to reflect this. Cheers, -- Joshua Colp Digiu

Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH

your verbose level is at a certain number or higher. It's nothing to be worried about. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.

Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH

t an error. Are you experiencing some issue that is causing you to think it is? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com &am

Re: [asterisk-users] core show channels verbose output

Mitch Claborn wrote: At the end of the output for "core show channels verbose" is a line that reads "4 active calls". Does anyone know how that number is formatted if there are more than 999 active calls? Will it have a comma or not? It will not have a comma. Cheers, -

Re: [asterisk-users] RTP IP re-write

of course. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Pr

Re: [asterisk-users] Motif/XMPP for Google Voice

Joshua Colp wrote: asterisk asterisk wrote: Dear all, Hola, I wish to ask a question of the new Motif Channel in asterisk 11. I successfully compile the binary and run without error. However, when dialing out, no external connection only ringing. During testing some issues were uncovered

Re: [asterisk-users] Motif/XMPP for Google Voice

channel driver, but unfortunately they did not make the last release candidate. My suggestion is to get Asterisk 11 from SVN or if you are not comfortable with that wait until the official Asterisk 11 release. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW

Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

ntioned. In the case of Asterisk it treats it as a SIP URI when promiscredir is enabled and doesn't use a peer entry, so it has no idea how to authenticate. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us ou

Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

authentication happen. I haven't tested that though. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.ast

Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

nfiguration option I mentioned. It won't hurt. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org --

Re: [asterisk-users] Motif XMPP

sues that have been discovered and expect that upon release it will work well for things I can control. I also personally will be extending it further so that in Asterisk 12 there will be additional features since it's a fun module to work on. What will they be? I don't quite know yet.

Re: [asterisk-users] motif load

sk remains in the media path but depending on the codecs this is optimized to be as efficient as possible. In the case of different codecs with transcoding occurring your performance hit would come from that, not from the fact you are using chan_motif itself. Cheers, -- Joshua Colp Digium, Inc.

Re: [asterisk-users] conversion?

, but is what i see correct (on a 1.8 system)? Yes, there is no capability for video transcoding in any version of Asterisk. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk

Re: [asterisk-users] Motif XMPP

ers in the Google network. I know the rules about cross-post and before casting stones - I've been around Asterisk and other platforms for a long time. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Ch

Re: [asterisk-users] Motif XMPP

ed before anyone can even come close to diagnosing your issue. To answer your question though there is nothing explicit you configure to have audio work. It should "just work". Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville,

Re: [asterisk-users] a=recvonly

d that to SIPADDHEADER or something? Asterisk doesn't allow you to manipulate the SDP as such so I'm afraid there is no way for you to change that value. Internally within applications they just discard the media received under such circumstances. Cheers, -- Joshua Colp Digium, Inc.

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