sudo
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Posts posted by sudo
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I would say 90 % this is a routing problem. The PBX is probably supposed to send the media to an address that it cannot reach, for example a hosted PBX sending to 192.168.1.100. You will also see this easily in the PCAP or the trace.
Can I send you the pcap in a PM? Im a little confused as there is no outbound rtp to see what IP its trying to send to.
Looking at the pcap, I only see the pbx ip, and my upstream provider. I would expect the rtp to be coming from the pbx ip, but alas, there is no outbound audio at all.
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Many appreciations
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Quick question.
On the general tab on the server it has the servers general stats.
I have a question regarding the "Calls" section. Is the number listed the total amount of call the system has completed in its whole life? Since the last reboot? For the past 24 hours?
Im having issues and im trying to determine how many calls we are getting in a 24 hour period. This is for the whole pbx, not per domain. Per domain would be nice too however.
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anything on the PCAP?
The pcap shows only inbound RTP
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I will check the router and get back to you.
Question - Even if the router was out of UDP sessions, wouldn't I still see the RTP stream attempting to go out. Wouldn't it be on the router that the stream gets dropped?
Im wondering why im not seeing the stream when the pcap was taken right off the PBX?
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Not sure if this is important, but something Ive noticed.
All of these phones are remote. This being the case, all phones on a domain use the same (external) IP to register. They are differentiated by port number.
Looks like some if not all of these phones are using ports in the RTP port range.
Im not sure if this info is pertinent. Hate to muddy the water...
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Thanks for that.
I should still be seeing the outbound RTP stream, correct? The pcap was from the server and I would expect to see the outbound RTP at least attempting to go out. But I do not see any outbound RTP from the server, just inbound from the outside party. This is confirmed in the pcap as well as I only hear the outside parties audio stream, and see only inbound RTP.
As far as the inbound RTP stream goes, it looks good. It hitting a good rtp port (57806) using ULAW. The server has only 1 IP address. The outbound proxy is configured on the trunk. I dont think its a firewall issue as it is not persistent. They call and it fails. They call again and it works.
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Im having a global issue where a number is dialed, the extensions phone counter starts, but there is no audio. When the call is disconnected, it disconnects.
SIP is going through fine, but not RTP. This is reflected in the PCAP.
During the call, there is no audio on either side, but it does ring the called party. The pcap has only the outside audio stream. In other words when I listen to the call, there is no audio coming from the internal extension, just the ringing and the called party answering saying, "hello...hello...CLICK"
As I stated, the pcap shows SIP going back and forward, but only inbound RTP. There is NO RTP coming out from the server.
The pcap was taken from the SnomONE server.
The inbound audio is hitting the right RTP port.
Im running Version: 5.0.10 (CentOS64)
Anyone have any ideas? Im at a loss and ive got angry customers.
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The issue was the timeout setting on the AA. It was set to 20 seconds. Modified it to 60 and it hits voicemail now.
Thanks snom ONE!
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If she is called directly, either internally or by DID it goes to voicemail. Just not when called through the AA.
I believe I sent you the logs in a PM with a call example. This was with all logs on 9.
Mailbox is enabled with default settings.
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Ive got an issue on an AA,
There is an option to dial the extension directly. This works fine. It rings the ext and they can answer it. The issue is if the call is not answered, it does not go to voicemail, it just drops.
This is happening on multiple extensions so I do not think its related to the ext settings, but the AA settings.
The 'Extension Input' under the Behavior setting on the AA are set to "When Extension matches'. Im not seeing any other settings that says what to do when voicemail is reached.
Any words of wisdom oh lords of the VoIP ;-)
~Sudo
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My guess would be that the service provider tries a re-invite and then something goes wrong.
You are correct. After 15 minutes the provider sends a reinvite then times out after 30 seconds and kills the call. Thats why its ~15:30
I modified the time out settings on the AA that this was hitting and it has cleared this issue up. Now im having a similar issue where they get to the general mailbox and it leaves a 5 minute long, blank message.
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It happened again last night. All the calls are 15:33 long. Does this number ring any bells for anyone?
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There is a service flag on the system. It is manually set so its hard to say if it was enabled when this call happened.
Is is set to forward to an outside number, but it is not the same calling number (669-600-XXXX), it is a local number.
Could something have happened that made the system forward back to the calling number?
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Ive got a cust that has a strange call example.
The call comes in and is redirected to itself. The call lasts 15+ minutes.
How/Why would it forward to itself? The call is hitting an IVR.
Here is what the call log looks like:
Time: 22:00
Dir: I
From: 669-600-XXXX
To: 1800470XXXX
Remote: +1669600XXXX (same as from)
Local:
Duration: 15:33This call took place on Sunday when no one was at the office to answer/transfer this call.
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Ive incresed the logging on all SIP, trunk, and general to 9.
One more interesting tidbit - The customer is complaining that when he gets to the office in the morning, the phones display is off and pixelated. He says that he has to restart the phone to get it back to normal. He has a PolycomSoundPointIP-SPIP_550 phone.
Does this trigger any thoughs as to what could be causing this issue? Ill be watching the logs and post what I see. I have heard complaints from customers on different domains on the same box so it appears to be a global issue. This happens on external as well as internal calls.
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Well according to the RFC, CANCEL cannot be sent before a provisional response (e.g. 100 Trying) has been received. This is not easy to implement, as the PBX then has to hold back the CANCEL until a response has been received; and keeping in mind that the response can be a 2xx class response, or even worse first a 18x response and then a 2xx response, the PBX has to then send the CANCEL and possibly also a BYE after this (actually it is also allowed just to send the BYE). It even gets worse, because all that stuff also depends on the transport layer, as TCP and TLS do not loose packets. Also keep in mind, that the PBX has to keep the transaction in memory for a long time, e.g. when there is absolutely nothing coming back from the other side.
Anyway, I would start with looking at the traffic that the PBX sends out on trunks. If the PBX really immediately sends out CANCEL after INVITE, we would have to check the reason why is that so. For example, if could be that the transaction timeout was explicitly set to 0 seconds, which would explain such a behavior.
Its within milliseconds that the cancel is sent out. This is confirmed from the PBX in a packet capture.
Where is the transaction timeout set? Can I send you a pcap to look over?
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I have an issue where the call drops after the the option is pressed on the IVR. In other words, the call is connected. They hear the IVR menu greeting, They press an option, 3 in this case, Extensoin picks up but hears nothing. The pcap shows that the calling party was still on the call. I can hear the calling party say "hello?" 10 seconds after the call is answered. No audio from the internal extensions side.
I know there is not much to go on here. What other info should I get? Anyone have a direction to point me in? The call flow looks good. The RTP is in the correct port range. Im really at a loss here...
Running Version: 5.0.10 (CentOS64)
Thanks in advance!
Sudo
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Running Version: 5.0.10 (CentOS64) and Im having issues on one domain.
I migrated this domain off an old pbxnsip box. They complained almost immediately that there was a degradation on call quality and call drops.
On my voip gateway I see these calls in the Failed Calls log with a 204 error. I called my upstream provider and they said that they see the INVITE come through then milliseconds later a CANCEL gets sent through prompting them to respond with a 481.
Thinking this could be a bug carried over from the old system, I completely rebuilt it from scratch with no avail. The problem persists and has even gotten worse.
What in the world could be causing this? The PBX? Their ISP? My gateway? In the wonderful world of VoIP im sure its all of the above.
I do not think its the gateway and Im leaning tward the pbx/domain.
I have not heard of this issue on any other domains.
Thank in advance.
Sudo
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Are you making sure the file being uploaded is a WAV file, 8 kHz Mono, 16 bit? Others won't be accepted.
I pulled them off the old systema and a backup of the current, so I would have to assuem they are in the correct format.
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I have accedentially deleted all the files in the /recordings/ dir. I was able to restore most but not all.
Im trying to upload the lost recordings from the WebUI. I go to the extensions mailbox tab, hit brouse, select audio, select save. After this there is no recording listed next to the greeting #. Just "No File Selected"
Im on ver. 5.0.10
Anyone know whats going on here? Why I cant upload audio? I was able to upload IVR greetings in this same domain fine.
I did import some .wav off the old servers (pbxnsip) /recordings/ directory to restore some audio files. Most of these domains were migrated off this old pbxnsip system. Perhaps the new system see's the same file and will not upload it? Its hard to tell which recording is which since theyre named att23.wav, name244.wav etc etc,
Thanks in advance!
Sudo
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I found the issue.
The DID was hitting a hunt group. The hunt groups From-Header was set to Group Name (Calling Number).
I set this to Calling party. I have not heard back from the customer to see if it worked, but Ive got a warm and fuzzy feeling that it will.
Thx
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Okay, So I was a little confused on what my customer was asking.
Apparently when they get an inbound call, the receptionist does not get the Name info of the DID, just the DID. But when she transferrs this to another extension, they get the full username and DID.
Any thoughs on why the recep is not getting this info
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Check out http://wiki.snomone.com/index.php?title=Trunk_Custom_Headers. Do you mean you need something like "ext-number" (no ANI)? Workaround could be to set a ANI that is the extension number. If you want to do that only for a trunk, you can put the trunk name in front of it, e.g. "2121234567 Trunk1:123".
I mean can we set it to display the user name along with the ANI. The First/Last name in the account.
No RTP coming out of SnomONE server
in General Setup
Posted
Ive sent it to you.
Thanks in advance.