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Steve B

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Posts posted by Steve B

  2. Raspberry PI 3

    in General Setup

    Posted

    That seems to be a problem with the right pbxctrl.dat file. Did you use version 5.4.1? I remember there was a version that somehow had the wrong web images.

     

    I have a trouble ticket for support to look at it and I will update this when they figure out what it is.

     

    Just curious though, is there a way to manually update the PBX from command line?

     

    Steve

  3. Raspberry PI 3

    in General Setup

    Posted

    So I tried my first install in on a Raspberry PI 3 as per the Raspberry install instructions.

     

    I am using the NOOBX installer of the current version of Raspbian.

     

    It seems to install ok and the pbx service is running.

     

    When I try to load the web page I get the welcome.htm but it is all in text no images. I attached screen cap of what I get. Is the PI 3 not supported yet or am I just missing a step?

     

    Thank you!

    Steve

    post-3310-0-80508600-1463462804_thumb.png

  5. I broke TLS

    in General Setup

    Posted

    Hi guys,

     

    I have been having a bit of an issue when it comes to one way audio through an incoming vitelity trunk on a TLS connected phone.

     

    So I was playing around with the ports and pow, no audio at all in TLS, I can get two way audio in UDP. And when I say no audio, no audio on the trunk or when dialing voicemail. If the call makes it to my cellphone, no problem. I have factory reset the PBX and no luck with audio in TLS, am I going to have to wipe the system and start again? Also, no audio from webRTC now either.

     

    I have changed routers to make sure this wasn't the issue. Still could be but UDP calls to and from the Vodia PBX and another PBX work flawlessly.

     

    The router's nat profile is port-restricted cone on both routers. I put the phone in the DMZ and no help there.

     

    Thanks,

    Steve

  6. 5.1.3

    in Announcements

    Posted

    We have made builds for all supported operating systems (Windows 32/64, Debian 32/64, CentOS 32/64, FreeBSD 64, Apple MacOS Intel/PowerPC, snom ONE mini) for the 5.1.3 release. The release notes can be found on http://wiki.snomone.com/index.php?title=Release_5.1 there are many fixes, we consider this release a major milestone regarding stability and features.

     

    Before you upgrade we recommend as usual to make a backup of the old installation. Also consider resetting your webpage changes or at least review them (webpages directory, html directory) as many things have been changes in the web interface.

     

    This version introduces a new firmware for snom phones. If you automatically provision snom phones, the phones will load a new firmware on the next reboot. Polycom phones that are running version 4 or higher will start using TLS and SRTP for phone calls after the next provisioning cycle.

     

    On webRTC when i make a call from the browser, am I missing where the keypad is or is that a work in progress? For instance if I call the voice mail, how do I enter a number :)

     

    You guys are making this thing hard to beat, keep up the good work.

  8. Version 5.1.0 has been released

    in Announcements

    Posted

     

    It is the plan to add more and more GrandStream devices. The challenge is to actually have the devices in-house for testing everything out. But maybe we can just use the 1400 as template and then make incremental changes. Let me know if you can help us testing it.

     

    I will see if I can get my hands on one to send back to you. I got it approved to send you a GXP2100 as long as we can get it back sometime. Let me know where to send it. I also have a GXW4104 gateway I can let you borrow, I know it is not a phone but if it is of use I will send it.

  9. Version 5.1.0 has been released

    in Announcements

    Posted

    I noticed Grandstream support for 140x, is there a possibility the 1450 plus the 21xx/22xx line may be coming soon. I have a job with these phones on it and the BLFs give me an issue. If a phone reboots, all the BLFs go blank until the corresponding extension or parking spot is used then the buttons work fine until the next reboot.

  10. PIKA WARP embedded appliance

    in Gateways

    Posted

    I'm going to be trialling a Sangoma Vega 50 soon to work alongside snom one for failover.

    I like the two Sangoma Vega 100 PRI appliances I have installed. They do not break, the only trouble ticket was when TLS broke on the PBX and I had to reboot it. The Patton FXO gateways have been good to me also.

  11. feature request: solicitors

    in General Setup

    Posted

    Serious?

     

     

    in the blacklist option.. can there be a setting to say "we dont accept solicicted calls, if you are not a solicitors press 1"

    kind of thing..

     

    db

     

    I think this is a fantastic idea, good way to weed out the solicitors from a legitment company calling in that is using a dialer.

  12. Public Beta

    in Announcements

    Posted

    I havent seen the release notes on this, does it address any problems with hunt group ringing in version 5.10? We are having issues where only certain phones will ring in a hunt group (all TCP) instead of all the ones listed but all the phones get the missed call text message.

  13. PBX adding a 1 to the 10 digit outbound CID

    in Call Treatment

    Posted

    First of all, I would set the country code in the domain to "1", so that the PBX knows that 10 digits are the same like 1 + 10 digits.

     

    On the trunk there is a setting called "Rewrite global numbers" where you can instruct the PBX to use a specific format for outbound calls. You can do this differently for each trunk. If you want to present only the 911 calls with 10 digits and all other with 11, you can for example set up two different trunks--one for 911 and another one for all other calls.

     

    Thank you. I have the country code set to 1 but anytime the PBX uses the extension ANI it sends 1+10 digit (extension ANI) as the caller ID even if i Have it set to 10 digit. What worked for me was making multiple outbound accounts liek you said, setting the Trunk ANI to the desired DID, and setting the from and remote party ID headers to Trunk ANI, worked perfectly. Thank you for your help.

  14. PBX adding a 1 to the 10 digit outbound CID

    in Call Treatment

    Posted

    I have a feeling this may be an easy one but need some help. I have my Snomone 5.10 connecting through a Vega 100 to a PRI with multiple rate centers. The Company I am working with has three locations, all with differnt 911 centers. When the PBX makes an outbound call either the PBX or the gateway is adding a 1 to the 10 digit CID. The carrier expects 10 digits on 911 calls to choose the right 911 center and defaults to the main BTN if it gets more. So my problem is that all 911 calls are ging to the default BTN because the carrier sees an 11 digit CID (they see this in their logs). Any help would be great, right now I am able to redirect calls to the Non emergency numbers until I can get this fixed.

  15. Cannot dial out to local cell phones

    in Gateways

    Posted

    Another problem on my Sangoma card is that as of last Friday we are not able to dial local cell phones from the Snom One plus. I can use my VoIP trunk to dial them and they are available cell phone to cell phone just not using the Sangoma Gateway. Other local numbers are accessible.

     

    If you disconnect the line cord from the card can you dial them from a regular analog handset? If so, have you checked the dial plan to see if the first few digits may be being picked up by another dialing pattern? If that is not the problem, does the log file show that the proper number is being sent to the card? That is where I would start, work form the physical side to the software side.

  16. apt-get update on SoHo

    in Linux-related Topics

    Posted

    The SoHo is not set up well. You have to keep in mind that the SoHo was supposed to be a training kit/giveaway. A proper Linux setup wasn't the goal.

     

    This comment makes me upset, not at you, but at snom. If this was a training kit / giveaway, it should not have been marketed as a small office home office appliance. I have one that I had to pull off my customers site that was about 14 months old because it quit responding on the network port and I had to replace it. That is a tough sell to a customer with a 1 year old phone system. I cringe at the thought of the other three that I have out and expect a call on those sooner or later as they are all approaching the same age. The idea wasn't bad, just the implementation of a small form factor mb with a 110vac to 12vdc transformer attached to the top of it in a confined case was probably not the best choice :) This is why I never bought any Mini's. It is more optimal for me to build a budget computer, install Linux, snomONE and your done. If one part goes bad its much cheaper than an entire system or waiting to send it back.

  17. New Features on 5.09

    in Announcements

    Posted

    Ok, I have had a chance to use the GUI for call handling! Everything works good and is awesome except for two problems. When I drag the call to a unregistered voice mail the system calls the person I transferred to the voice mail back when they hang up and only gives the caller dead air and doesn't seem to connect it to an extension. The second thing is if the person I transferred hangs up in the middle of the transfer, the same thing happens.

     

    I haven't been able to check this using a snom phone yet, only a grandstream so that could possibly be a problem. Maybe someone can test this in a snom phone environment. When the call is transferred, to the voicemail or a parking spot I get a busy signal on the headset.

     

    Other than that, I love it!

     

    Oh never mind, I think I figured out the problem. My customers voip provider, recalls the incoming number if the remote party hangs up and the PBX is still waiting for media. I bet the system is holding the leg open longer than the voip provider needs to redial. I can see 1 instance where this would be nice to redial customers but other than that its a lame feature on the providers part.

  18. New Features on 5.09

    in Announcements

    Posted

    Ok, I have had a chance to use the GUI for call handling! Everything works good and is awesome except for two problems. When I drag the call to a unregistered voice mail the system calls the person I transferred to the voice mail back when they hang up and only gives the caller dead air and doesn't seem to connect it to an extension. The second thing is if the person I transferred hangs up in the middle of the transfer, the same thing happens.

     

    I haven't been able to check this using a snom phone yet, only a grandstream so that could possibly be a problem. Maybe someone can test this in a snom phone environment. When the call is transferred, to the voicemail or a parking spot I get a busy signal on the headset.

     

    Other than that, I love it!

  19. Suitable Hardware

    in PBX Lounge

    Posted

    Hi,

     

    I just built the first unit using the 350 as the chassis, an Atom D2500 as the core. and a small SSD for storage. It is exactly what I was anticipating. Does the job, looks presentable, it is affordable and allows you to say the catchy phrase: All solid state, no moving parts.

     

    Now then. Here at my hometown I could also get (for about the same price) an ECS NM-70 I2 mother board that has a Celeron dual core 847 processor (same form factor). It would be more speed everywhere (processor, RAM and even access to the hard drive). Being price about "equal" the pros and cons I see are the following: In favor of the Atom; 1) Three years warranty as opposed to only one, 2) Fan-less processor (one less thing that could go wrong) and in favor of the Celeron: 1) Raw speed (The processor scores 1042 against 407 on pass-mark benchmark, RAM runs at 1333 MHz as opposed to 1066 and Access to the SSD is at 6 Gb per second instead of only 3)

     

    Being cost practically the same, it is only a thing of valuing reliability against speed. One way of thinking would be that since electronic parts tend to either fail pretty soon or last for years, perhaps I should favor the faster solution and stress test the units before sending them to a final user. The other way of thinking would be to use the Atom for users without plans for growth and the Celeron for the ones that are either large already or have plans for growth. The problem here is that I do not have an idea of how many concurrent calls would each system be able to handle.

     

    I would appreciate your thoughts and insights.

     

    The Celeron 847 processor is only 1.1 ghz vs the atom's 1.87 ghz. I see the front side bus difference but what makes the cerleron faster? I am not being sarcastic, I really want to know.

  20. Vitelity Registrations

    in Trunk Setup

    Posted

    Well when registration fails, the system has already re-sent the REGISTER messages many times (500 ms, 1 s, 2 s, 4 s, 8 s, 8 s, 8 s). Giving it a "break" for 60 seconds does not sound unreasonable to me. You also have to keep in mind that too many registrations may be DoS for the registrar. The key question here is why the registration fails so often. If other trunks stay up all the time that is a hint that the internet connectivity is stable (which is usually the problem); the only think that I can think of is that the registrar expects the registration on a different IP address for fail-over or load balancing purposes, which causes problems. Also you have to keep in mind that the DNS server might also play a critical role in this game. In order to rule that out I would out an IP address in the outbound proxy to see if that could be the issue.

     

    I finally got to it, here is the log file, the funny thing is is that the PBX gets a 200 OK back from Vitelity then says the trunk is 408:

     

    May  3 16:41:51 BestPBX Trunk 22: Preparing for re-registration 
May  3 16:41:51 BestPBX Trunk Vitelity Outbound: Sending registration to 64.2.x.x 
May  3 16:41:51 BestPBX Resolve 2738182: udp 64.2.x.x 5060 
May  3 16:41:51 BestPBX SIP Tx udp:64.2.x.x:5060: REGISTER sip: SIP/2.0#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-d9f3be5373cd55817229d26d4ec47834;rport#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88015 REGISTER#015#012Max-Forwards: 70#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;+sip.instance="<urn:uuid:f282f982-9f3c-4c3e-91f6-bb4682171d9b>"#015#012User-Agent: snomONE/4.5.0.1090 Epsilon Geminids#015#012Supported: outbound#015#012Expires: 3600#015#012Content-Length: 0#015#012#015
May  3 16:41:51 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-d9f3be5373cd55817229d26d4ec47834;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88015 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Content-Length: 0#015#012#015
May  3 16:41:51 BestPBX Message repetition, packet dropped 
May  3 16:41:51 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-d9f3be5373cd55817229d26d4ec47834;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>;tag=as04681bd6#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88015 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29c9acdc"#015#012Content-Length: 0#015#012#015
May  3 16:41:51 BestPBX Answer challenge with username **myusername** 
May  3 16:41:51 BestPBX Resolve 2738183: udp 64.2.x.x 5060 udp:1 
May  3 16:41:51 BestPBX SIP Tx udp:64.2.x.x:5060: REGISTER sip: SIP/2.0#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-15acf27157a93b1e9491d99cd18899c1;rport#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88016 REGISTER#015#012Max-Forwards: 70#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;+sip.instance="<urn:uuid:f282f982-9f3c-4c3e-91f6-bb4682171d9b>"#015#012User-Agent: snomONE/4.5.0.1090 Epsilon Geminids#015#012Supported: outbound#015#012Authorization: Digest realm="asterisk",nonce="29c9acdc",response="67a94278dd46790b5a437d1c89c79b8a",username="**myusername**",uri="sip:",algorithm=MD5#015#012Expires: 3600#015#012Content-Length: 0#015#012#015
May  3 16:41:51 BestPBX Message repetition, packet dropped 
May  3 16:41:52 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-15acf27157a93b1e9491d99cd18899c1;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88016 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Content-Length: 0#015#012#015
May  3 16:41:52 BestPBX Message repetition, packet dropped 
May  3 16:41:52 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-15acf27157a93b1e9491d99cd18899c1;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>;tag=as04681bd6#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88016 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Expires: 60#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;expires=60#015#012Date: Fri, 03 May 2013 23:02:39 GMT#015#012Content-Length: 0#015#012#015
May  3 16:41:52 BestPBX Trunk 22: setup callback to send re-registration after 30 seconds 
May  3 16:41:53 BestPBX Clearing outbound proxy for trunk 22 (Vitelity Outbound) 
May  3 16:41:53 BestPBX Registration on trunk 22 (Vitelity Outbound) failed with code 408. Retry in 60 seconds 
May  3 16:41:53 BestPBX Trunk status Vitelity Outbound (22) changed to "408 Request Timeout" (Registration failed, retry after 60 seconds) 
May  3 16:41:53 BestPBX DNS: Request mail.**maildomain**.com from server 24.166.x.x 
May  3 16:41:53 BestPBX DNS: Add CNAME mail.**maildomain**.com **maildomain**.com (ttl=14400) 
May  3 16:41:53 BestPBX DNS: erasing AAAA mail.**maildomain**.com, id 9855 retry count 0, 
May  3 16:41:53 BestPBX DNS: Request **maildomain**.com from server 24.166.x.x 
May  3 16:41:54 BestPBX DNS: Add AAAA **maildomain**.com  (ttl=60) 
May  3 16:41:54 BestPBX DNS: erasing AAAA **maildomain**.com, id 9856 retry count 0, 
May  3 16:41:54 BestPBX DNS: Request **maildomain**.com from server 24.166.x.x 
May  3 16:41:54 BestPBX DNS: Add A **maildomain**.com 69.89.x.x (ttl=14400) 
May  3 16:41:54 BestPBX DNS: erasing A **maildomain**.com, id 9857 retry count 0, 
------
May  3 16:42:53 BestPBX Trunk 22: Preparing for re-registration 
May  3 16:42:53 BestPBX Trunk 22: sending discover message for  
May  3 16:42:53 BestPBX Resolve 2738188: discover 64.2.x.x 
May  3 16:42:53 BestPBX Trunk 22: Received reply for discover method 
May  3 16:42:53 BestPBX Trunk 22 (Vitelity Outbound) is associated with the following addresses: 64.2.x.x 
May  3 16:42:53 BestPBX Trunk Vitelity Outbound: Sending registration to 64.2.x.x 
May  3 16:42:53 BestPBX Resolve 2738189: url sip:64.2.x.x 
May  3 16:42:53 BestPBX Resolve 2738189: udp 64.2.x.x 5060 
May  3 16:42:53 BestPBX SIP Tx udp:64.2.x.x:5060: REGISTER sip: SIP/2.0#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-08db4856e9b1ff185c6580d17c56c9f8;rport#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88017 REGISTER#015#012Max-Forwards: 70#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;+sip.instance="<urn:uuid:f282f982-9f3c-4c3e-91f6-bb4682171d9b>"#015#012User-Agent: snomONE/4.5.0.1090 Epsilon Geminids#015#012Supported: outbound#015#012Expires: 3600#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-08db4856e9b1ff185c6580d17c56c9f8;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88017 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX Message repetition, packet dropped 
May  3 16:42:53 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-08db4856e9b1ff185c6580d17c56c9f8;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>;tag=as0fcb37c9#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88017 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="291177d1"#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX Answer challenge with username **myusername** 
May  3 16:42:53 BestPBX Resolve 2738190: udp 64.2.x.x 5060 udp:1 
May  3 16:42:53 BestPBX SIP Tx udp:64.2.x.x:5060: REGISTER sip: SIP/2.0#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-e6721577bd050373f62092372d8b8669;rport#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88018 REGISTER#015#012Max-Forwards: 70#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;+sip.instance="<urn:uuid:f282f982-9f3c-4c3e-91f6-bb4682171d9b>"#015#012User-Agent: snomONE/4.5.0.1090 Epsilon Geminids#015#012Supported: outbound#015#012Authorization: Digest realm="asterisk",nonce="291177d1",response="b3933bbcffb57db488bf9fa64d764143",username="**myusername**",uri="sip:",algorithm=MD5#015#012Expires: 3600#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX Message repetition, packet dropped 
May  3 16:42:53 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-e6721577bd050373f62092372d8b8669;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88018 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX Message repetition, packet dropped 
May  3 16:42:53 BestPBX SIP Rx udp:64.2.x.x:5060: SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 24.119.x.x:5060;branch=z9hG4bK-e6721577bd050373f62092372d8b8669;received=24.119.x.x;rport=5060#015#012From: "Vitelity Outbound" <sip:**myusername**@>;tag=1864809879#015#012To: "Vitelity Outbound" <sip:**myusername**@>;tag=as0fcb37c9#015#012Call-ID: eixyvo4y@pbx#015#012CSeq: 88018 REGISTER#015#012User-Agent: packetrino#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO#015#012Supported: replaces#015#012Expires: 60#015#012Contact: <sip:**myusername**@24.119.x.x:5060;transport=udp;line=b6d767d2>;expires=60#015#012Date: Fri, 03 May 2013 23:03:41 GMT#015#012Content-Length: 0#015#012#015
May  3 16:42:53 BestPBX Trunk status Vitelity Outbound (22) changed to "200 OK" (Refresh interval 30 seconds) 
May  3 16:42:53 BestPBX Trunk 22: setup callback to send re-registration after 30 seconds

     

    I have the PCAP also, just have to go get it off the server.

  22. 401 error on SIP registration

    in Trunk Setup

    Posted

    I am not sure by your response, if you left the actual url as "pbx.company.com" but unless you added an alias ie yourcompany.localurl.com, to the ip of internet connection, it wont work. The registrant will not be able to find the url pbx.company.com as it doesnt exist.

    Did you leave the registration username off on purpose or is it really not there.

  23. Vitelity Registrations

    in Trunk Setup

    Posted

    The PBX re-registers depending on what it is being told by the registrar. However you can override that (did you already?) with the "Keepalive Time". Then you can be 100 % sure this is not a problem with a too short registration interval.

    I understand that part, I am talking about when the trunk loses registrations, such as a 408 error (I shouldn't have used the 401 error above in my description). It would be nice for the trunk to have an option of immediately re-registering and having a setting to be able to say how often to try and re-register after that. If we have to wait 60 seconds to re-register, this can/has lead to support calls. I have another system that allows us to set this under the SIP settings.

  24. Vitelity Registrations

    in Trunk Setup

    Posted

    Some providers have said IP registration which is just an IP authentication that I have set up as a proxy server. Are you monitoring your SIP registrations with Vitelity to see when they are going up and down. I will try the in and and out and see if the registration is more reliable than before. I would get a lot of 408 errors when I was running a SIP registration. Thank you, I will report back a bit later and tell you how it is working.

    My connection to Vitelity went 408 unreachable 64 times between 12:30 yesterday and 8:00 AM this morning. The rest of my trunks stayed up without a problem. Is there a reason snomONE takes 60 seconds to re-register instead of an immediate re-register on 408, 401 then every 10 seconds until it comes back up. A minute can be an eternity on some systems. At least let us be able to make the decision how we want it to re-register.

  25. Vitelity Registrations

    in Trunk Setup

    Posted

    Steve,

    When you say that here's your "IP registration" set-up; what do you mean by that, exactly? There're two ways to connect a trunk to Vitelity: (i) SIP Registration or (ii) IP authentication. If you're willing to tinker a little bit, you may be able to get rid of that 10-second delay by employing the workaround I described in a different forum topic (see post #12). If you follow my example, you'll want to make sure that you change the "type" variable on each trunk from "proxy" to "registration." In addition, you'll need to log into your customer portal at Vitelity and create a login ID & password for a "Sub Account" under the DIDs tab.

     

    Some providers have said IP registration which is just an IP authentication that I have set up as a proxy server. Are you monitoring your SIP registrations with Vitelity to see when they are going up and down. I will try the in and and out and see if the registration is more reliable than before. I would get a lot of 408 errors when I was running a SIP registration. Thank you, I will report back a bit later and tell you how it is working.

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