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jlumby

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Everything posted by jlumby

  1. It means that the PBX received a new media stream on the RTP port. Based upon the setup of your gateway/ITSP, this can be a very normal thing.
  2. I have been noticing that certain situations cause a long delay in the audio between two parties, and I was wondering what may cause this. It seems that the more legs that are involved in a call, the longer the delay gets. For example, if an extension calls an extension on another PBX that is directly networked to it, then the call is usually fine, however if it forks to the cell phone, then there is a couple second delay. However however if that same extension calls the cell directly (utilizing the same PSTN gateway, just not forking through the second PBX, then there is no noticeable delay) Since it does not happen with every call, I am wondering if adaptive jitter buffers are being additive. Most all of the links in this example have a fairly consistent latency of ~ 50ms, however I have seen occasions where the latency might jump to 120ms for one instant within the total duration of the call. Does this cause the jitter buffer to increase, and never decrease again for the duration of the call? We are not having any noticeable audio issues such as cutting out, dropping, or 1 way audio. Just looking for some input as to how I can fix the long delay that does not happen all the time.
  3. One of our customers softswitches got hit by a DoS attack this morning. I am attaching the packetcapture from before I blocked it at the firewall. It ran the processor up to 99% and the memory up to 1 gig. After blocking the IP, it took stopping/starting the service to reclaim the memory. Just want to make sure that the newer versions will automatically protect against attacks like this. The customer was running 3.3.2.3183 (Win32) DoS.zip
  4. I was wondering what features are coming down the line to protect from SIP based DoS attacks against PBXnSIP. I am concerned because on 2 different occasions I have had a PBX go down (99% cpu utilization on pbxctrl.exe) because of malformed registration packets. The packets were caused from a router that did not properly work with SIP. I am worried because this was unintentional, so I could imagine the impact malformed registration packets could have if someone was intentionally trying to make the server unusable
  5. Is there a way that when an extension is setup to forward that it can send the ANI of the extension, and not the caller that is being forwarded. In general I like it sending the original caller's ID, however I have 2 extensions that I would like to send the extension ANI so that the after hours answering service can determine which company it is coming from.
  6. That document was exactly what I was looking for. The second part of the request was to be able to use a variable to insert one of the parameter fields into the CDR record. I am looking for a way that I could put the company name into the CDR data. I have a situation where I am hosting multiple companies in the same building. I need to keep them all as part of the same domain so that they can call each other directly by extension number, however I would like the CDR data to show the company name. I do not want to enter the company name into the first, or last name field since it would be visible in the caller ID. I was thinking that i could input the company name into the Parameter 1: field under the account's registration tab, however i would need a variable that would allow me to pull this value into the CDR output.
  7. There used to be documentation that stated what all of the CDR format variables were, and I can no longer find it. It seems like it has turned into a document that simply states the different locations you can write the CDR data to. Does anyone know of a current document that lists the format variables? As for the second issue, I would like to know if there is a variable out there that I can use to read one of the extension's parameter fields, and insert it into the CDR output.
  8. I was wondering if there was a way in the sip world that after an attended transfer that the caller ID could be updated to show that person that is currently connected, and not the initial caller. I was not sure if a re-invite or something like that would do the trick.
  9. Does anyone know of a phone that supports conferencing more than 3 people together? I have a client that does not want to utilize the conference bridge, and would like to be able to see all parties on the display of their phone, and selectively drop one.
  10. Well, I do not use the autoprovisioning that is built into PBXnSIP, probably just because I have been using the phones long before they could autoprovision them. I have a base config that I got off of my callmanager, and then I manually modify it to match each phone. As for The SRV record. In simple terms it is a DNS record that tells the client (phone) who is hosting the service they are looking to connect to, which order to try them in, and which port they are listening on. Basically it just points to the basic DNS A record(s), which then resolve to an IP address. As for your DNS issues, I would be happy to PM back and forth with you, since there are tons of different ways you can handle the inside VS outside IP address. The simplest/least expensive, which works well if you are not going to be moving the phones from the inside to the outside of your network frequently is to create 2 different DNS records for your domain. Since you can have multiple aliases in PBXnSIP for your domain, this is not an issue. FOr example if your domain name is voip.yourdomainname.com then I would create the following records A records: voip.yourdomainname.com - resolves to your public IP address voipp.yourdomainname.com - resolves to your private IP address SRV Records: _sip._udp.voip.yourdomainname.com - Lists voip.yourdomainname.com on port 5060 _sip._udp.voipp.yourdomainname.com - Lists voipp.yourdomainname.com on port 5060 In PBXnSIP create your primary domain name as: voip.yourdomainname.com Also list the following aliases for the domain: voipp.yourdomainname.com voipp voip The last 2 are because the Cisco phone does not append the domain suffex in the regrestration packet. Then in the config file for the phones that are going to be on the inside of your network I would set the extension button proxy to voipp, and the ones sitting on the outside of your network I would set to voip Since this is all additions to what you already have setupo, none of this will have any effect on the SNOM phones
  11. I am using the Cisco phones in multi domain mode without any issues. Here is the key to making it work, and keeping the phone stable at the same time. The first thing is make sure you have the SRV records created for the domain. The second thing is the outbound proxy needs to be empty. In the extension proxy field, ONLY type in the dns host name of the PBXnSIP server, and not the FQDN of the server. The reason is the phone will append the DNS suffex it receives from the DHCP server. So obviously you need to make sure your DHCP server is handing out the proper DNS suffex. Even though the phone appends the DNS suffex to resolve the IP of the server, it does not append it in the registration packet, and therefore you need to create a domain alias on the PBXnSIP that is just the host portion of the FQDN. If you do all that, I am sure it will work properly.
  12. It changes to whatever the minimum time is. That seems to be a major problem. Why is the PBX sending that out?
  13. I am having an issue with several different products registering to PBXnSIP, mostly Audiocodes, Grandstream, Quickphones, and Adtran. The following is a packet capture of an Audiocodes gateway. When I set the registration expiration to 3600 on the audiocodes (as you can see in the packet capture below), the PBX replies with 29 I am not sure why this is. There is no nat going on between the gateway and the PBXnSIP, as well as other devices such as cisco phones do not have this problem, and they successfully register for the full 3600 to the same domain on the same switch. I am very concerned about such short registration time, since when I once had all of my extensions set to register every 60 this high frequency caused processor spikes that led to poor call quality. Now that I have the majority of my extensions set for 3600 the spikes, and quality issues have instantly gone away. I have even noticed that this same problem exists when registering one PBXnSIP to another. My current PBXnSIP version is 3.3.2.3183 (Win32) From AudioCodes to PBXnSIP: No. Time Source Destination Protocol Info 11 0.612851 10.0.0.131 10.0.0.6 SIP Request: REGISTER sip:voipp.twincitytelephone.com Frame 11 (832 bytes on wire, 832 bytes captured) Ethernet II, Src: AudioCod_18:a8:e9 (00:90:8f:18:a8:e9), Dst: Supermic_83:f6:cb (00:30:48:83:f6:cb) Internet Protocol, Src: 10.0.0.131 (10.0.0.131), Dst: 10.0.0.6 (10.0.0.6) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:voipp.twincitytelephone.com SIP/2.0 Method: REGISTER [Resent Packet: False] Message Header From: <sip:213@voipp.twincitytelephone.com>;tag=100eb030-8300000a-13c4-50029-a-215d8afe-a SIP from address: sip:213@voipp.twincitytelephone.com SIP tag: 100eb030-8300000a-13c4-50029-a-215d8afe-a To: <sip:213@voipp.twincitytelephone.com> SIP to address: sip:213@voipp.twincitytelephone.com Call-ID: 100fc348-8300000a-13c4-50029-a-2a94a760-a CSeq: 2 REGISTER Sequence Number: 2 Method: REGISTER Via: SIP/2.0/UDP 10.0.0.131:5060;rport;branch=z9hG4bK-b-2b4a-2a9b4f24 Transport: UDP Sent-by Address: 10.0.0.131 Sent-by port: 5060 RPort: rport Branch: z9hG4bK-b-2b4a-2a9b4f24 Max-Forwards: 70 Supported: replaces,100rel Allow: REGISTER, INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, PRACK, SUBSCRIBE Expires: 3600 Authorization: Digest username="213",realm="voipp.twincitytelephone.com",nonce="afa4b325771a1d526dd8f81c14701937",uri="sip:voipp.twincitytelephone.com",response="a9e3feb7a512a2ec7762275f25a30797",algorithm=MD5 Authentication Scheme: Digest Username: "213" Realm: "voipp.twincitytelephone.com" Nonce Value: "afa4b325771a1d526dd8f81c14701937" Authentication URI: "sip:voipp.twincitytelephone.com" Digest Authentication Response: "a9e3feb7a512a2ec7762275f25a30797" Algorithm: MD5 Contact: <sip:213@10.0.0.131:5060> Contact Binding: <sip:213@10.0.0.131:5060> URI: <sip:213@10.0.0.131:5060> SIP contact address: sip:213@10.0.0.131:5060 User-Agent: MP202 B 2FXS/2.6.4_p5_1_build_9 Content-Length: 0 From PBXnSIP to AudioCodes No. Time Source Destination Protocol Info 12 0.623656 10.0.0.6 10.0.0.131 SIP Status: 200 Ok (1 bindings) Frame 12 (458 bytes on wire, 458 bytes captured) Ethernet II, Src: Supermic_83:f6:cb (00:30:48:83:f6:cb), Dst: AudioCod_18:a8:e9 (00:90:8f:18:a8:e9) Internet Protocol, Src: 10.0.0.6 (10.0.0.6), Dst: 10.0.0.131 (10.0.0.131) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 Ok Status-Code: 200 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 10.0.0.131:5060;rport=5060;branch=z9hG4bK-b-2b4a-2a9b4f24 Transport: UDP Sent-by Address: 10.0.0.131 Sent-by port: 5060 RPort: 5060 Branch: z9hG4bK-b-2b4a-2a9b4f24 From: <sip:213@voipp.twincitytelephone.com>;tag=100eb030-8300000a-13c4-50029-a-215d8afe-a SIP from address: sip:213@voipp.twincitytelephone.com SIP tag: 100eb030-8300000a-13c4-50029-a-215d8afe-a To: <sip:213@voipp.twincitytelephone.com>;tag=1a11ed798c SIP to address: sip:213@voipp.twincitytelephone.com SIP tag: 1a11ed798c Call-ID: 100fc348-8300000a-13c4-50029-a-2a94a760-a CSeq: 2 REGISTER Sequence Number: 2 Method: REGISTER Contact: <sip:213@10.0.0.131:5060>;expires=29 Contact Binding: <sip:213@10.0.0.131:5060>;expires=29 URI: <sip:213@10.0.0.131:5060> SIP contact address: sip:213@10.0.0.131:5060 Date: Tue, 21 Jul 2009 22:16:12 GMT Content-Length: 0
  14. I have found out that the problem only exists if the PAC is running at the same time that you try to place the TAPI call. Do you know of a workaround?
  15. That is correct, the ANI in the extension field will take precedence over all, and the ANI on the domain will take precedence over the trunk
  16. What I have done is setup a global dialplan that routes all 911 calls to a specific trunk. You can then fill in the ANI field with a format like 1111111111 E911trunk:2222222222 What this does is sends the ani of 1111111111 if the call goes out any trunk, except if the dialplan sends it out the trunk named E911trunk, then it will send an ANI of 2222222222
  17. Have a situation where I setup the huntgroup "From Header" to Group Name (Calling Party) The issue I am having is logically I would expect that the output on the phone would read: Group name (Caller ID Name) Caller ID number. The problem is it shows: Group Name (Caller ID Number) Caller ID Number. It seems illogical that the PBX would populate the parenthesis with the Caller ID number, and not the Caller ID Name since the number is duplicated in the next field. Can you shed some light on this, or create the more logical version as an option in the next release?
  18. I believe it is network/computer related I tried connecting to the same server from a different computer at a different location, and it works fine. I will be taking my known good laptop back out to the customer's location tomorrow to narrow it down to the network. We did not design their data side of the network, so I am leaning towards that router as the problem.
  19. I am running a system with xp and outlook 2003 with the tapi driver 1.1 The PBX is 3.3.2.3183 (Win32) When I click to place a call, the phone rings once, and before i can pick it up, it stops. I have the client's XP firewall set to allow TCP, and UDP 5060-5062. I have no problem dialing the exact same number directly from the phone. I am seeing the following in the log file [6] 2009/07/15 16:23:51: TAPI: Click to dial to 14193922384 [5] 2009/07/15 16:23:52: INVITE Response 503 Not Implemented: Terminate 753f1452@pbx [5] 2009/07/15 16:23:52: Not setting dialog state of non-existing call port (call-id=84732f57@pbx#62701) [5] 2009/07/15 16:23:52: INVITE Response 487 Request Cancelled: Terminate 4fa8d6d9@pbx
  20. I am looking for the voicemail users guide. I cannot seem to find it after they got rid of the wiki
  21. So can you turn loopback protection back on, and call from one domain to another utilizing Try Loopback in the dialplan? In theorey if you are correct, it should work since it would only be one single loop.
  22. I am just thinking statistically it would be more common for a user to accidentally create a loop back to their own domain, and since the average user cannot get to the dialplan, we would assume someone with knowledge would be editing the dialplan. Therefore if it utilizes the Try Loopback in the dialplan it could be trusted, and if it looped internally, or through the phone company it should not be trusted. It would be neat if there was a TTL for the loopback protection. This way it could be turned on, however it would kill any loop after a preset number of hops.
  23. I have had some success, however I am concerned that loopback detection needs to be turned off. Is there a way to keep loopback detection on to protect from external loops, however trust the dialplan?
  24. The real issue is that I am using 3 different manufacturers hardware, and all 3 of them do not support it. It is not as common of a feature as it is made out to be. The grandstream ATA will only subscribe to it's own extension. I have found the same thing with an Audiocodes ATA I had in stock, as well as the Cisco 7961 phones. I think that when 3 different manufacturers all cannot do it, then it is not a real viable solution
  25. I have a situation where I have 2 of the owners homes tied to the main office. The problem I am having is that he would like one mailbox for all of his messages that is accessable from all phones at both of his homes (plus at the office). The problem is they are a mix of phones, including several running off of analog ATA ports where the only indication is stutter dialtone. In order to accomplish this, with the mix of phone technology, I had originally thought of registering them all as the same extension, however this causes a problem transfering from one room of the house to another, as well as since there are 2 homes, I have to indicate different E911 locations, and I cannot do that if they are all the same extension. So now I am wondering if there is a way to point all of the voice mail boxes of the different extensions to one common mailbox, and that mailbox would activate the MWI on all of the different extension numbers?
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