Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Hi, I do not have a switch to mirror the traffic... I am only remotely connected to the office, where all is set up. I have full control over asterisk and the phone and I tcpdumped the traffic coming from the phone. The weird thing is... if I configure the SIP-Server Setting to use TCP on Port 80, I see REGISTER requests. If I configure to use UDP only on Port 5060, I do not see nothing at all... not a single Request coming from the phone... and, yes... It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... Meanwhile I tried all available firmware-Versions, with and without provisioning. I am wondering about downloads, the phone is trying to receive from downloads.polycom.com that constantly fail (yes, these files do not exists there, the phone can communicate with the internet...) On the other hand, I donĀ“t think that this has something to do with the problem, as the phone tries to REGISTER when I use TCP / 80 Olivier, would you mind and mail me your config-files and some screenshots from the phone-webconfig? Which software-versions are you using? thank you, yves if someone wants to take a look at the phone-logs: boot-log 02.335|so |*|01|-- Initial log entry -- 02.335|so |*|01|+++ Note that Updater log times are in GMT +++ 02.335|boot |*|01|Initial log entry. Current logging level 3 02.335|copy |*|01|Initial log entry. Current logging level 3 02.335|utilm|*|01|Initial log entry. Current logging level 4 02.335|hw |*|01|Initial log entry. Current logging level 4 02.335|ethf |*|01|Initial log entry. Current logging level 4 02.335|dns |*|01|Initial log entry. Current logging level 3 02.335|curl |*|01|Initial log entry. Current logging level 3 02.335|sec |*|01|Initial log entry. Current logging level 4 02.641|wdog |*|01|Initial log entry. Current logging level 4 02.641|lldp |*|01|Initial log entry. Current logging level 3 02.641|cdp |*|01|Initial log entry. Current logging level 3 02.641|key |*|01|Initial log entry. Current logging level 4 02.642|so |3|01|Platform: Model=SoundStation IP 6000, Assembly=3111-15600-001 Rev=W Region= 02.642|so |3|01|Platform: Board=3111-15600-001 B 0 02.642|so |3|01|Platform: MAC=0004f2070cd3 02.643|so |3|01|Platform: BootBlock=3.0.4.0001 (15600-001) 11-Jul-12 08:53 02.644|so |*|01|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56 02.644|so |3|01|Application, main: Label=Updater, Version=Azurite 5.0.5.2324 09-Dec-13 15:31 02.644|so |3|01|Application, main: P/N=-Y-YYY 02.644|log |*|01|Install file upload callback for 'Updater' 02.644|app1 |*|01|Initial log entry. Current logging level 3 02.645|cfg |*|01|Initial log entry. Current logging level 2 02.651|app1 |3|01|Application, load: Type=SIP, Version=4.0.4.2906 18-Apr-13 01:11 02.652|boot |*|01|Using TFFS for flash load 02.652|boot |*|01|Code length: 0x0097A585 02.652|boot |*|01|Code checksum: 0x4B86ABFB 03.631|so |3|01|Link status is Net up Speed 100 full Duplex. 17.497|app1 |4|01|Loaded application sip.ld from local system successfully. App-log 001139.870|app1 |*|03|Manual Reboot 001139.870|so |5|03|soAudioChannel compiledOffsetsApply error: unrecognized verAudio 11 for headset 001140.026|so |*|03|SoNcasC::procMsg: Client service shutdown complete 001144.025|wdog |*|03|Watchdog Expired: tSup 04.975|log |*|03|-- Initial log entry -- 04.975|so |*|03|Platform: Model=SoundStation IP 6000, Assembly=3111-15600-001 Rev=W Region= 04.975|so |*|03|Platform: Interfaceeth0 MAC=0004f2070cd3 04.977|so |*|03|Platform: BootBlock=3.0.4.0001 (15600-001) 11-Jul-12 08:53 04.977|so |*|03|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56 04.977|so |*|03|Platform: Updater=5.0.5.2324 09-Dec-13 15:31 04.977|so |*|03|Application, main: Label=SIP, Version=Mink 4.0.4.2906 18-Apr-13 01:11 04.977|so |*|03|Application, main: P/N=3150-11530-404 04.977|rdisk|*|03|RAM disk created, size: 8,388,608 bytes 04.978|ocsp |*|03|O.C.S.P. Enabled = 0 04.978|tls |*|03|Initial log entry. Current logging level 4 04.998|pmt |*|03|Initial log entry. Current logging level 4 04.998|wdog |*|03|Initial log entry. Current logging level 4 04.998|ethf |*|03|Initial log entry. Current logging level 4 04.998|hw |*|03|Initial log entry. Current logging level 4 04.998|ares |*|03|Initial log entry. Current logging level 4 04.998|dns |*|03|Initial log entry. Current logging level 4 04.998|cfg |*|03|Initial log entry. Current logging level 4 04.998|dot1x|*|03|Initial log entry. Current logging level 4 05.000|cfg |*|03|RT|Network eth0 link went up 05.000|cfg |*|03|RT|cfgRtNetInterfaceUpdate
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- 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] Polycom SoundStation IP 6000 does not register
Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- 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] pjsip realtime - endpoints not loading.
We are continuing to test our asterisk 13 pjsip deployments. I am running into an issue that I am assuming is a configuration problem, and am hoping someone can point me in the right direction. We are running pjsip in real-time mode using a database to store all the endpoint records. Our endpoint records with our carrier do not support registration. The issue I am having is when asterisk starts none of the non registration endpoints become available. They will not allow calls inbound or acknowledge qualify's. To get them to come on line we have to do a pjsip show endpoints, and then all works until asterisk is restarted. Is there any way to get the endpoints to load without manually doing the pjsip show endpoints? Any input is appreciated. Thanks Bryant -- _ -- 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] Polycom SoundStation IP 6000 does not register
On Wed, Dec 21, 2016 at 7:50 AM, Yves wrote: > Hi Mark, > > yes, you are right... these are different VLANs > I configured the other phone to use the same IP (192.168.1.13)... and it > worked flawlessly... on the SAME Networkcable in the same plug... > so it must have something to do with the polycom phone config... remember... > when I use tcp the phone tries to register, but does not even try with > udp... > > thank you, > yves > I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? > > Am 21.12.2016 um 13:34 schrieb Mark Wiater: > > Yves, > > Didn't you say that > > AsteriskServer: 192.168.1.211 > SIP-user: 165 > > ? > > On 12/21/2016 4:24 AM, Yves wrote: > > . It is sure for 100% that there is no firewall or something else mangeling > in between... another Hardphone works as expected using the same > Netzworkcable on the same Networkplug with UDP on Port 5060... > > > This other hardphone, what IP does it have? > > > 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask > 255.255.255.0 > > The line above suggests to me that your phone and your asterisk server are > on a different network, there has to be something that routes between those > two networks. Often what routes, can firewall. > > 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 > Temporarily not available > > > > Mark > > > > > -- > _ > -- 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] Polycom SoundStation IP 6000 does not register
sorry... typo the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: On Wed, Dec 21, 2016 at 7:50 AM, Yves wrote: Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- 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] Polycom SoundStation IP 6000 does not register
Hi Yves, Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the phone. Maybe with the snom this not happen because your switch don't see the MAC of the Snom as a "supperted IP Phone". 2016-12-21 13:59 GMT-03:00 Yves : > sorry... typo > the problematic phone has the 192.168.0.13 > the asterisk has 192.168.1.211 > > when i connect a snom phone on the cable that was in the soundstation 6000 > before and configure the > phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... > > it would be helpful if someone, that has a running soundstation ip 6000 > could send the configuration... :-/ > > regards, > yves > > > > Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: > >> On Wed, Dec 21, 2016 at 7:50 AM, Yves wrote: >> >>> Hi Mark, >>> >>> yes, you are right... these are different VLANs >>> I configured the other phone to use the same IP (192.168.1.13)... and it >>> worked flawlessly... on the SAME Networkcable in the same plug... >>> so it must have something to do with the polycom phone config... >>> remember... >>> when I use tcp the phone tries to register, but does not even try with >>> udp... >>> >>> thank you, >>> yves >>> >>>I am a bit confused: is your problematic phone's IP 192.168.0.13 >> (what the error log is reporting below) or 192.168.1.13? >> >> Am 21.12.2016 um 13:34 schrieb Mark Wiater: >>> >>> Yves, >>> >>> Didn't you say that >>> >>> AsteriskServer: 192.168.1.211 >>> SIP-user: 165 >>> >>> ? >>> >>> On 12/21/2016 4:24 AM, Yves wrote: >>> >>> . It is sure for 100% that there is no firewall or something else >>> mangeling >>> in between... another Hardphone works as expected using the same >>> Netzworkcable on the same Networkplug with UDP on Port 5060... >>> >>> >>> This other hardphone, what IP does it have? >>> >>> >>> 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask >>> 255.255.255.0 >>> >>> The line above suggests to me that your phone and your asterisk server >>> are >>> on a different network, there has to be something that routes between >>> those >>> two networks. Often what routes, can firewall. >>> >>> 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 >>> Temporarily not available >>> >>> >>> >>> Mark >>> >>> >>> >>> >>> -- >>> _ >>> -- 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 > -- GnuPG Key ID: 0x39BCA9D8 https://www.github.com/mefhigoseth ...:::[ God Rulz ! ]:::... -- _ -- 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] I think this is a bug (video call file) 11.25.1 and 13.13.1
On 12/20/2016 06:01 PM, Jerry Geis wrote: >Hi Jerry, > just had a look through the code, and from what I can tell, what >you're trying to do is not supposed to work, exactly. It appears that >what Asterisk expects is to be given a filename, such as "myplayback". >Asterisk will first search for an audio version of the file (like >myplayback.gsm or myplayback.opus), and open that as an audio stream. If >that succeeds, it then will also see if there is an accompanying video >stream (such as myplayback.h264). If it then finds that video, then the >result will be that Asterisk will play the audio from the audio file and >the video from the video file. >What this means is that Asterisk does not properly handle: >* Files that have audio and video streams contained within >* Video files without accompanying audio >This is one of those times where Asterisk's handling of video is not >user-friendly and in general ass-backwards and terrible. If you have a >tool that can extract the audio to its own file, then you would be able >to run your scenario, presumably. >It would be a welcome addition for Asterisk to be able to open a single >file containing video and accompanying audio and be able to play those back. Hi Mark, Thanks for your reply... I just tried what you suggested on only got audio. I created a wav file and put it in the /tmp directory just like the video.h264 file. So /tmp has video.h264 and video.wav both. I then placed the call and only heard the audio from the wav file. I used this for my call file: Channel: SIP/2002 Context: testing Extension: 99 Priority: 1 Application: Playback Codecs: h263,h264,vp8,g722,ulaw,alaw,wav Data: /tmp/video My Bria 4 softphone uses the h263 and h264 codecs and of course wav file audio. Based on your look of the code did I miss something to trigger the playing of the video file? I can extract the audio out to a seperate file - so not a show stopper for me. No errors showed up on the Asterisk CLI when I did my test. Thanks so much, Jerry I don't see anything obvious in the code that would have prevented the video from playing back. Unfortunately, the debug from Asterisk isn't going to be especially helpful here, with one exception. If you have core debug at level 1 or higher, then when Asterisk detects the video file, it will say: "Ooh, found a video stream, too, format h264" If you see that message, that at least means that Asterisk is finding the video file as expected. If you don't see that, then it's likely that Asterisk is unaware of the h264 file format type. It may be that you don't have the format_h264.so module loaded. It may be that there was an error that occurred when that module was loading, causing it not to be able to load properly. If you are seeing that debug message, it at least means that Asterisk attempted to play back the video file, but something else in the process caused the video not to play back as expected. The first thing you could check is packet captures to see if Asterisk is even attempting to send video to the softphone. If so, then it's likely that there is some sort of codec mismatch happening (likely something in the format parameters). If Asterisk is not even attempting to send any video, then it likely means that there is some other issue. It may be a bug, or it may be some erroneous condition in the environment. Hard to tell yet though. Mark Michelson -- _ -- 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 realtime - endpoints not loading - Solved
It appears that res_odbc.so does not always load fast enough to allow the realtime mappings in the extconfig.conf to complete successfully at startup thus stopping the first load of the pjsip endpoints and other pjsip values. The resolution for this is to preload the res_odbc.so and res_config_odbc.so in the modules.conf. The realtime mappings then appear to complete correctly during startup and allows all the pjsip data to load correctly. Thanks Bryant Sent: Wednesday, December 21, 2016 9:12 AM Subject: [asterisk-users] pjsip realtime - endpoints not loading. We are continuing to test our asterisk 13 pjsip deployments. I am running into an issue that I am assuming is a configuration problem, and am hoping someone can point me in the right direction. We are running pjsip in real-time mode using a database to store all the endpoint records. Our endpoint records with our carrier do not support registration. The issue I am having is when asterisk starts none of the non registration endpoints become available. They will not allow calls inbound or acknowledge qualify's. To get them to come on line we have to do a pjsip show endpoints, and then all works until asterisk is restarted. Is there any way to get the endpoints to load without manually doing the pjsip show endpoints? Any input is appreciated. Thanks Bryant -- _ -- 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