similar to: How to make channel busy signal?

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume is very low (maybe a few per week), but we have multiple offices in three geographic locations and would like to be able to email the images to the correct location. For planning purposes,

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

I'm about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? Thanks, -Dave -------------- next part

Does anyone have a Sipura 3k running, and using the FXO? I've got things working right, but if I try to toss a *67 in the dialplan, it seems the sipura is throwing a 403 forbidden back. For example: exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/${EXTEN:1} works fine exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/*67${EXTEN:1} does not (even if I toss a couple Ws in) I can't

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

I'm just curious why Sipura isn't using free IAX protocol with their devices instead of SIP? With IAX NAT traversal would have been easier, so why are they using SIP. Is there any politics in it? -- #Joseph

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

All- I'm attempting to set up a Sipura SPA-3000 in Sao Paolo, Brazil. Not being a portuguese speaker, I'm having a rough time of finding the relevant information on how to make the thing pick up the PSTN line and make an outbound call. The sipura in question works fine on a bench connected to a POTS line in the US, but is now plugged in in Sao Paolo. The immediate thing I notice is that

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What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option

>> Has anyone benchmarked Asterisk on a dedicated single versus dual >> processor machine? > > http://www.astertest.com/ > > Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more than twice as fast as non trunking