I am having problems with my ATA (Sipura 1001, Firmware 3.1.8, hardware 2.0.1). The scenario is:
1. The caller places a call successfully, and then places the called party on HOLD.
2. The Called Party then hangs up.
3. The call will remain On Hold until you either open the wire from your telephone to the VOIP box, or until you take the call off hold and hang up your phone.

When you do this same thing with a phone connected to a PSTN line, the phone disconnects after staying on hold for approx. 10 seconds.

Any ideas on how to make this disconnect?

We're running software version 3.1.8(SEc) on hardware version 2.0.1(3953) and have the following observations:

1) To use G726 with asterisk 1.2.6 or later you must edit the rtp.c and either dfine USE_DEPRECATED_G726=1 or remove the #ifdef USE_DEPRECATED_G726 statement. If you do not do this you will be able to place calls using the SPA but not receive calls. This is because the SPA identifies the G726 codec using a deprecated paylode type.

2) In the "Line" settings there is an option for "Use Preferred Codec Only". If this is set to "No" (the default) and you try and create a second call instance by pressing flash and dialing a new number the second call will only use ulaw/alaw. As far as I can tell, the only way to fix this is to set the "Use Preferred Codec Only" to "Yes". We're not sure why this happens but it does and if you do not have the correct allow= statement in your sip.conf you will not be able to use three-way or conference calling.

On another note, the combination of the STUN server and other settings make this device a very attractive, inexpensive device. We just bought 10 of them and even without the provisioning tool (which we can't seem to get from Linksys) they don't take long to setup.

Since I don't like high-bandwidth codecs (ulaw/alway) for voice traffic, I highly recommend re-compiling asterisk with the deprecated G726 option and using that codec. G729 is also an option, but since the license keys are tide to the sum total of the ethernet MAC addresses I refuse to use it. And if you want to support three-way calling, the only way to force a lower-bit-rate codec is to use only one codec (the preferred one) I choose G726. After setting it up and re-compiling it works great.

Since we use these in a corporate environment, we have also changed the "Dial Plan" to "(89,x.T|*01x.T|*02x.T|*xx|1-7xxx)". 8* and 9* are all external numbers, the *-codes work and 1000-7000 are internal extensions.

The one other very helpful hint that I found was to set the "VMWI Serv" to "No", which will disable the short ring of the phone when a voicemail message is received.

We also changed the SIP Registration timeout to 300 seconds for some of our devices that run on remote networks - it seemed to solve problems with in-bound call handling.

We have also configured one device to work with TelIAX and ulaw with high-jitter-buffer for faxing, which works well when network traffic is low.

We're running software version 3.1.8(SEc) on hardware version 2.0.1(3953) and have the following observations:

1) To use G726 with asterisk 1.2.6 or later you must edit the rtp.c and either dfine USE_DEPRECATED_G726=1 or remove the #ifdef USE_DEPRECATED_G726 statement. If you do not do this you will be able to place calls using the SPA but not receive calls. This is because the SPA identifies the G726 codec using a deprecated paylode type.

2) In the "Line" settings there is an option for "Use Preferred Codec Only". If this is set to "No" (the default) and you try and create a second call instance by pressing flash and dialing a new number the second call will only use ulaw/alaw. As far as I can tell, the only way to fix this is to set the "Use Preferred Codec Only" to "Yes". We're not sure why this happens but it does and if you do not have the correct allow= statement in your sip.conf you will not be able to use three-way or conference calling.

On another note, the combination of the STUN server and other settings make this device a very attractive, inexpensive device. We just bought 10 of them and even without the provisioning tool (which we can't seem to get from Linksys) they don't take long to setup.

Since I don't like high-bandwidth codecs (ulaw/alway) for voice traffic, I highly recommend re-compiling asterisk with the deprecated G726 option and using that codec. G729 is also an option, but since the license keys are tide to the sum total of the ethernet MAC addresses I refuse to use it. And if you want to support three-way calling, the only way to force a lower-bit-rate codec is to use only one codec (the preferred one) I choose G726. After setting it up and re-compiling it works great.

Since we use these in a corporate environment, we have also changed the "Dial Plan" to "(89,x.T|*01x.T|*02x.T|*xx|1-7xxx)". 8* and 9* are all external numbers, the *-codes work and 1000-7000 are internal extensions.

The one other very helpful hint that I found was to set the "VMWI Serv" to "No", which will disable the short ring of the phone when a voicemail message is received.

We also changed the SIP Registration timeout to 300 seconds for some of our devices that run on remote networks - it seemed to solve problems with in-bound call handling.

We have also configured one device to work with TelIAX and ulaw with high-jitter-buffer for faxing, which works well when network traffic is low.

I am well am having frustrating fax issues. I am getting intermittant failures (mixed with successes) using the SPA-1001. Latest firmware, settings as suggested by Sipura. They don't have phone support so I am running out of luck (and time). Anyone else out there have suggestions?

Has anyone gotten fax to work well over this ATA?

I finally had the opportunity to get my SPA-1001 incorporated into my pbx. It took me all of 5 minutes, a TA reboot, and I was up and running (minus some time from assigning an extension to it). The sound quality is better than what I have with Vonage.