similar to: ATA 186 Codec Question.

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Clicking on the "For more information, click here" link on the Digium site nice brings back up the same page I was looking at before, without any additional G.729 information that I can see. I'm wondering if some kind asterisker out there could provide us neophytes with some "typical scenarios" where that codec would be useful to us. For instance, I assume that it

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

Hi All, I'm looking at setting up a Asterisk system, and hope to use ATA 186's with it. Im in Australia, and am getting mixed answers to if its the I1 or I2 i need, does anyone have any experience with using ATA 186's in Australia Also, can anyone recommend a good place to obtain these locally? Cheers, Steven

I have 2 Asterisk servers connected with IAX. It's working fine I can call an extension from one phone in an office to another phone in the other office. The only problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow

Does anyone know how long the orders take? I ordered some a couple of days ago and it said normally 24hours, and I am guessing that the weekend cause's some delays but it did not say anything abouy that. Any one got any ideas on how long generally over the weekend it takes? Thanks David

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

The details for the Australian cisco ATA186 are below: > -----Original Message----- > From: Tony Du [mailto:tony.du@action.com.au] > Sent: Tuesday, 8 July 2003 4:31 PM > To: 'Adam Goryachev' > Subject: RE: [Asterisk-Users] ATA 186 in Australia > > Hi Adam, > > I sold a Cisco ATA186 I1 2 port adaptor (Cisco code: > SW-SMH-UL-ATA-2P)to you on 16/10/02) >

Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

Ok I have a question. Seen it come and go around the mailling list for a while but never really seen an answer that seems to sort it out. What is needed is some interface from * > Mobile Phone > Mobile Network Service. At this point all the providers in AUS that I have found are charging a Premium Rate for Land Line > Mobile Network services. What I would like to do is be able to

Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Hi, I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use g.723) Asterisk will connect to iConnect, successfully natively bridge the call and then about two seconds later not just drop the call, but terminate unexpectedly.

> Message: 1 > Date: Fri, 21 Jan 2005 17:38:27 -0600 > From: "Henry Devito" <hdevito@qwest.net> > Subject: [Asterisk-Users] SPA-2000 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn> > Content-Type: text/plain; format=flowed;

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)