similar to: DTMF and SIP

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP

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

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]:

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

Hi I just wanted to know if anyone has done the following, or knows how to. When a customer dials into *, we would then ask them to for an account number (which they would type in with the key pad), we would then ask them to select and option (1 to 9). We would then like to capture this information and send it in an email. All hints 'n tips welcome... -------------- next part

Hi I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can use all 4 lines for out going calls fine, but on incoming we can only use 2. On calling in using the main msn, the 3rd line gives a an engaged signal. I have unplugged 1 of the cards, and the other card takes the 2 calls. I then swapped this around, and this also works fine. But when using both cards, we can only use

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

Can't get asterisk to understand DTMF from x-lite. Used proposed configuration on the web. Still doesn't work. Using inband dtmfmode, still no go. Help? Vmail.cgi doesn't work as well, error says "Premature end of script headers: vmail.cgi" Shoval -------------- next part -------------- An HTML attachment was scrubbed... URL:

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Hi! I've been trying to use the Voicemail (and Voicemail2) applications with an SIP Phone (X-Lite, for those who cares), when I use g.711(a/u) codec, it works perfectly with inband (it detects the whole mailbox (in my case 10007)), but not with rfc2833 (in this case, it only detects 107 as the mailbox number). With gsm codec, the inband doesn't work, I guess that's due to the

Hello people, I'm trying to run a cobol application under wine and I'm getting several problems with forms windows. Some forms windows in my app are appearing larger than the screen size (800x600) but the wm don't let me resize then to see all the form. I already try setting the font size to a smaller size but windows still get larger than the screen. I also tried changing from GNOME

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

Hello All, Yesterday, one of our customer's cluster(2 box) server goes down after a I/O error during the backup procedure. When we turn on the cluster again, the 600M database file have been lost, disapear. Now, I'm with one of the disks (the other was used to rebuild the database server) and need to know how can I recover any piece of data from the disk. After the crash we mount the