similar to: channel.c switches to gsm even when sip.conf only allows ulaw

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

Is anyone else with ViaTalk experiencing an outage right now? My DID has been down since 5AM (8/20). Asterisk is unable to re-register or connect for outbound calls. I have also tried calling support and their number gives a fast busy.

I just put in a new Asterisk@Home 2.8 system. Trunk is connected via SIP to ViaTalk. I had an older Asterisk@Home system up and running that was working fine and I replicated settings over to the new box. When I call 7777 from an internal SIP extension I can hear the IVR menu just fine. However, when I call from a POTS phone to our number and it comes in via ViaTalk over SIP the call connects

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965XXXX context=viatalk-in type=user

Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob.

asterisk-users-request@lists.digium.com is believed to have said: >Hi, > >Sorry for being very late on this thread but i am trying to make a >decision on which one to go for. Options are > >1. Dock n Talk offered by Voxilla (USD139) >2. GSM Gateway by CyberTelecom (GBP60) > >I'm having a TDM400P with 1 FXO & FXS. > >I'm interested in implementing DISA

Hi, Sorry for being very late on this thread but i am trying to make a decision on which one to go for. Options are 1. Dock n Talk offered by Voxilla (USD139) 2. GSM Gateway by CyberTelecom (GBP60) I'm having a TDM400P with 1 FXO & FXS. I'm interested in implementing DISA my Asterisk@Home. Need some recommendations for experienced guys out here... Thanks... Dan

Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and

Does anyone on-list know of a serice provider that can provide DIDs in 713-861-xxxx? I'd like to port my AT&T POTS lines to an IP based service into my Asterisk box. Michael

ulaw blank spots but gsm fine We've got plenty of QoSd bandwidth to run ulaw. Yet when we do, we get drops (blank spots) in the call. GSM codec eliminates the blanks, but is of course much less quality. Any ideas as to what would be causing this? Is ulaw less loss tolerant than gsm? I've run tests using iperf, and I'm looking at maybe 10 per 50000 packets are lost. (around 0.02%).

I'm looking for the best recording I can get of Allison saying "Steve" or "Ian". I found gsm recordings of both out there but was looking for something higher quality. Can anyone point me in the direction of a WAV or ULAW recording of those names? Thanks in advance Steve

> From: "John Hughes" <john at calva.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Sent: Thursday, May 14, 2020 2:10:45 AM > Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and > alaw; asterisk wants to translate g729 -> alaw. WHY? > I am having a

Hello, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 > or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw > right?  In fact from the sip debug it looks like it does, but then I > get the dreaded "channel.c:5630 set_format: Unable to find a codec > translation path: (g729) -> (alaw)"