similar to: Asterisk, inband DTMF send by a GSM mobile

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.

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

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

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

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

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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

When I make a call over the Internet and call myself IN over POTS my phone rings to outside party but I can not hear it. Why isn't my channel extension indicating busy status when I'm making call over Internet? This way I could ring my next extension with n+101 priority. I'm using Sipura-3K unit. -- #Joseph

>> 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

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

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,

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Hi all, I just upgraded a production server to asterisk 1.4.25, compiling with the following: [*] 1. DONT_OPTIMIZE [*] 2. DEBUG_CHANNEL_LOCKS [*] 3. DEBUG_THREADS [*] 4. DEBUG_FD_LEAKS [ ] 5. LOW_MEMORY [*] 6.

We are running HEAD from last night and 1.0.5 and 1.0.3 and 1.0.2 and they all are running just fine in production environments each handling thousands of calls a day. I suppose reliability depends upon what you are using, but for our purposes they all are very stable. I could do without the memory leaks though. MATT--- -----Original Message----- From: Florian Lefeuvre

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

Hello, currently I am solving the problem of having GSM gateway - can you please document how any of you solved the Asterisk to GSM thing? I would appreciate if you write what HW did you use and what was the price of it, pros and cons. I have limited budget, but I'm opened to every idea. Thank you very much. Marcel