I wanted to trigger a voip call, so I used one Ubuntu system where I have installed ‘asterisk server’ another Ubuntu system I have used where two different types of softphone ‘microSIP’ and ‘Twinkle’ have installed.
When I message from one phone to another it is successfully accepted by the n/w. But problem is that not able to trigger a call from one to another softphone. It’s instantly rejected the call although I set 10 sec hold timing,when other user not accepting the call it will be dropped after 10 sec only.
I have added the error screen shot for your reference -
Assistance your guidance please -
Please provide logs as plain text, taken from the log files. Preferably enable /var/log/asterisk and use that.
You have multiple problems although some are hidden under the screen glare.
You are using an obsolete and unsupported channel driver.
You have no compatible codecs configured. In fact the other side appears to be refusing all your codecs.
You have an excessive number of codecs enabled. This can cause problems because of packet size limitations. You should disallow all and then allow the one or two that you actually need.
You have a serious problem with AstDB, which I’ve never seen before.
Hi all,
Right now I am getting authentication failure message , in that case I am running microSIP softphone on the host system and asterisk server on the container with the same host system .
Like host ip: 10.138.x.x - this one using for microSIP
Container ip: 192.168.x.x - where asterisk server running here
Please suggest what need to do here-
Thanks & Regards
Saddam
Check your firewall if need be turn your firewall off while you register your sip phone then restart your firewall back up.
Provide logs and configurations as plain text, taken directly from the files, and not rekeyed. A photograph of a screen is even more difficult to handle than a screenshot.
Also, note that chan_sip is no longer supported, and few people here know much about it.
I would, however, take
at face value.
Hi all,
I understand but there is security restriction even screen shot is not allowing, so I am sharing the photocopy only.
Please let me what would be the solution-
Thanks
On Fri, May 31, 2024, 4:13 PM david551 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
david551
May 31Provide logs and configurations as plain text, taken directly from the files, and not rekeyed. A photograph of a screen is even more difficult to handle than a screenshot.
Also, note that chan_sip is no longer supported, and few people here know much about it.
I would, however, take
at face value.
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Hi tareqmy and all,
I am running microSIP softphone on the host system (ip:10.138.x.x) and asterisk server on the docker container called ‘ext-dn’ on the same host system only.
docker_oai: 192.168.70.x
Ext-dn : 192.168.70.x
But getting registration failure msg ,where for sip client configuration I am giving domin = container ip(Ext-dn) here.
Please suggest is there I am doing anything wrong.
I have added sip client configuration details in the below
Thanks & Regards
Saddam
Hi, try to tick “Allow ip rewrite” in microsip client.
Hi,
I did it but no luck.
I think you have to post sip trace of registration.
Without is very difficult to investigate.
Regards
Hi,
Yes I got it , but there is a security restricted , I am sharing wireshark log screen shot, please check once and let me know.
Thanks
On Fri, May 31, 2024, 4:13 PM spady7 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
spady7
May 31I think you have to post sip trace of registration.
Without is very difficult to investigate.
Regards
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In text mode, please.
Hi all,
Good morning!
Right now I am facing RTP packet routing issues over the user interface.
VoIP_Client_1 : 10.138.77.X1
User interface: 12.1.1.X
VoIP_Client_2: 10.138.77.X2
Asterisk_Server: Docker container (192.168.70.X3)
SIP packets I am able to routing over the user interface 12.1.1.X but RTP packet communicating through host(77.X1) to host (77.X2) where asterisk server running inside the docker container on the host system 77.X2.
How to route RTP packets like SIP?
Thanks
On Fri, May 31, 2024, 7:22 PM spady7 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
spady7
May 31In text mode, please.
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Hi all,
Good morning!
Right now I am facing RTP packet routing issues over the user interface.
VoIP_Client_1 : 10.138.77.X1
User interface: 12.1.1.X
VoIP_Client_2: 10.138.77.X2
Asterisk_Server: Docker container (192.168.70.X3)
SIP packets I am able to routing over the user interface 12.1.1.X but RTP packet communicating through host(77.X1) to host (77.X2) where asterisk server running inside the docker container on the host system 77.X2.
How to route RTP packets like SIP?
Thanks
On Tue, Jun 4, 2024, 12:14 PM saddam hossain <sh40713@gmail.com> wrote:
Hi all,
Good morning!
Right now I am facing RTP packet routing issues over the user interface.VoIP_Client_1 : 10.138.77.X1
User interface: 12.1.1.X
VoIP_Client_2: 10.138.77.X2
Asterisk_Server: Docker container (192.168.70.X3)SIP packets I am able to routing over the user interface 12.1.1.X but RTP packet communicating through host(77.X1) to host (77.X2) where asterisk server running inside the docker container on the host system 77.X2.
How to route RTP packets like SIP?
Thanks
On Fri, May 31, 2024, 7:22 PM spady7 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
spady7
May 31In text mode, please.
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I know little about Docker, but what I do know is that greatly complicates Asterisk configuration. Its use is discouraged.
To the extent that this is not Docker related, you should disable direct media.
Hi Divid,
Ok, I’ll check with that.
Thanks
On Tue, Jun 4, 2024, 5:29 PM david551 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
david551
June 4I know little about Docker, but what I do know is that greatly complicates Asterisk configuration. Its use is discouraged.
To the extent that this is not Docker related, you should disable direct media.
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Hi David,
I checked it and right now it showing RTP routing between 192.168.70.1 to 192.168.70.145, that communication looks like softphone and asterisk server running on the same host , earlier it was host to host ip start from 10.138.
But still is not not routing over the user interface.
Thanks.
On Tue, Jun 4, 2024, 6:00 PM saddam via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
saddam
June 4Hi Divid,
Ok, I’ll check with that.Thanks
On Tue, Jun 4, 2024, 5:29 PM david551 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:
david551
June 4I know little about Docker, but what I do know is that greatly complicates Asterisk configuration. Its use is discouraged.
To the extent that this is not Docker related, you should disable direct media.
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What does the Asterisk RTP log show?
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