 Thread: HP Ipaq Voice Messenger 514 |
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I have an HP Ipaq 514 that I understand has sip functionality. However I am unable to get it to register to pbxes. The VoIP config for the phone is done via a configuration wizard. Here are my settings:
It searches but on the phone's display it says pbxes: No Service.
Does anyone have any ideas? Does anyone have experience with the SIP functionality on HP Iqaqs?
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I fixed this by setting up another trunk for vitelity for incoming calls and changing the SIP server setting to inbound4.vitelity.net
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I have several DIDs through Vitelity that until recently were working perfectly. I haven't made any changes to my configuration, but a few days ago I started getting errors like the one below:
A call to your DID 2295164604 has failed at 11:20am on 02/16/2008. We received 'CHANUNAVAIL' when attempting to route the call to your server or device.
This error usually means your server or device is not currently registered to our servers. Please make sure you are registered to the correct inbound server for your account as per the configuration samples found on the support page within the user portal and that you have selected the correct routing method for your DID(s) on the DIDs page.
I also notice that all of my DIDs ring busy and don't make it to pbxes voicemail like they should be. However, I can place outgoing calls via the Vitelity trunk and see the traffic on both pbxes and vitelity CDRs. When I look at the status screen in pbxes i see all items blinking green then red continuously. I tried clearing cache and refreshing the browser window, but that didn't change anything.
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My mobile provider (MTS Russia), charges per minute for incoming calls from outside Russia. If I set up a call forward to my mobile phone in PBXes, can I make it appear to MTS that the call is originating from within Russia even though the call is actually coming from a US phone number? My SIP provider is in the US as well.
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I have a premium account at PBXes and I want to get the most out of it but I need help. I have an idea of what I want to accomplish and I want to pay i-p-tel help me to do it, but I don't know the best way to do this. I think it is more involved than a normal support issue. My proposed scheme involves acquiring some US DIDs but am not sure what exactly I need, so I need advice as well as configuration help.
Most of my family and friends are in the US and Germany, but I am in Moscow Russia. Here is my ideal scheme:
Can somebody help?
Thanks!
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| Thread: RE: Fax |
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Can the separate, dedicated DID for fax be from Sipgate?
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| Thread: RE: Fax |
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I should mention that my DID is from sipgate. I will try to find out if sipgate supports G.711.
What is a vocoder?
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| Thread: RE: Fax |
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I can't receive faxes either. I've had one DID that I receive calls on no problem. I set the extension of fax machine to system and put my email address there, but when I try to send a fax to my DID, my phone rings and eventually I get a voicemail message of the fax. Why is the system not recognizing a fax? Is there something I should do that I haven't? Can I use my existing DID or do I have to have a separate one?
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I had some settings that were off. I made changes and now I am registering successfully to pbxes! The changes I made included:
1. Enabled STUN
2. Changed SIP Remote-Party-ID to No
3. Changed Refer-To Target Contact to Yes
4. Changed Make Call Without Reg to Yes
5. Changed DNS SRV Auto Prefix to No
6. Changed Use Pref Codec Only to No
7. Changed DTMF Tx Method to AVT
8. Changed Enable IP Dialing to Yes
I'm not sure which of these were actually causing the problem though.
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Thanks for the direction. Now everything seems to be working well from SJPhone on my computer, however I can't seem to be able to get my SPA-3102 to register successfully to pbxes.org. I'm using the same credentials that I am using successfully with SJPhone. I also tried setting up a second extension to use with the ATA and that didn't work either. Any ideas?
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I want to try to use PBXes with my SPA-3102, but I don't have any idea how to get my ATA registered. Is there any documentation available to help with? I have my ATA registered to voicestick and when I tried to create a sip trunk for that, I got the following error:
"Cannot connect to Asterisk Manager with /
This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available."
I sure this is because I don't know what I am doing.
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