James Mahood Posted September 10, 2008 Report Posted September 10, 2008 When I register the trunks with the new ViaTalk servers I can no longer places calls from one trunk to another. This hasn't been a problem with the old ViaTalk servers. There appears to be a compatibility issue between version 2 and version 3 of PBXnSIP software and the new ViaTalk servers. The Invite comes into the server but no extensions ring. ViaTalk plays a recording that the party is unavailable. Quote
Vodia PBX Posted September 10, 2008 Report Posted September 10, 2008 When I register the trunks with the new ViaTalk servers I can no longer places calls from one trunk to another. This hasn't been a problem with the old ViaTalk servers. There appears to be a compatibility issue between version 2 and version 3 of PBXnSIP software and the new ViaTalk servers. The Invite comes into the server but no extensions ring. ViaTalk plays a recording that the party is unavailable. Well, they send the INVITE of the PBX back to the PBX - and the PBX responds with "Loop Detected". Not sure why they send the request back to the PBX, including a lot of Record-Route and Via headers. Quote
James Mahood Posted December 10, 2008 Author Report Posted December 10, 2008 You must turn off Loopback Detection in the General settings for PBXnSIP to work with ViaTalk. With Loopback Detection off you can also dial your own number and your other phones will ring. If Loopback Detection is on you will hear rining but the other phones won't ring. Quote
James Mahood Posted December 20, 2008 Author Report Posted December 20, 2008 Simultaneous ring to a mobile phone has one way audio most of the time even with PBXnSIP version 3.1.1.3110. This issue is even worse if you have a virtual number and someone calls the virtual number and you answer on your mobile. In that case you will always get one way audio. Another issue with PBXnSIP and ViaTalk's new servers is if you have another device on you network like and IP phone or a soft phone directly registered to ViaTalk and you call their support number you will always get one way audio. Calls to other number and your ViaTalk trunks work fine. Beware ViaTalk is getting very uncooperative if they think they are trouble shooting PBX issues. My last issue they insisted that I shut off the PBX and register my line to a soft phone. Only then would they run a trace or do any testing of my one way audio issue when answering a ViaTalk Simultaneous ring call on my mobile phone. Unfortunately for us they were right. The problem vanished when the trunk was registered to a soft phone. Quote
hosted Posted December 31, 2008 Report Posted December 31, 2008 http://www.nexSIP.com I emailed you a test trunk group to try out earlier today. Quote
Vodia PBX Posted December 31, 2008 Report Posted December 31, 2008 Simultaneous ring to a mobile phone has one way audio most of the time even with PBXnSIP version 3.1.1.3110. This issue is even worse if you have a virtual number and someone calls the virtual number and you answer on your mobile. In that case you will always get one way audio. Another issue with PBXnSIP and ViaTalk's new servers is if you have another device on you network like and IP phone or a soft phone directly registered to ViaTalk and you call their support number you will always get one way audio. Calls to other number and your ViaTalk trunks work fine. Beware ViaTalk is getting very uncooperative if they think they are trouble shooting PBX issues. My last issue they insisted that I shut off the PBX and register my line to a soft phone. Only then would they run a trace or do any testing of my one way audio issue when answering a ViaTalk Simultaneous ring call on my mobile phone. Unfortunately for us they were right. The problem vanished when the trunk was registered to a soft phone. The problem is that ViaTalk obviously uses the SIP proxy model as interface to the customer. That has a couple of problems. First, you can see the Record-Route headers, which is a open door for DoS of their service and a interop nightmare because of all the devices out there that have problems with strict and loose SIP routing. The next problem is that SIP UDP packets can easily get bigger than 1492 bytes, many routers cut additional bytes of the UDP packet off. Maybe the softphone ignoes that and thats the reason why it works with the soft phone but not the PBX. I don't blame ViaTalk for being uncooperative. It is simply impossible to troubleshoot such complex problems. Let me say it again, session border controllers are worth every single cent! Quote
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