similar to: Sudden audio loss

Hello, We're running Asterisk 1.8.14.1 and our carrier requires us to send a 200 OK for OPTIONS request in order for them to keep sending traffic to our endpoints. Asterisk is currently replying with 404 messages, and their SBC only accepts 200 OK responses. How do I configure asterisk to reply with 200 OK without changing any source code? Regards, Grant -------------- next part

On Tue, Feb 17, 2015 at 5:14 AM, Grant Bagdasarian <gb at cm.nl> wrote: > > Hello, > > > > We?re running Asterisk 1.8.14.1 and our carrier requires us to send a 200 > OK for OPTIONS request in order for them to keep sending traffic to our > endpoints. > > Asterisk is currently replying with 404 messages, and their SBC only > accepts 200 OK responses. >

The problem: Strangely, the PDC rejects user password challenges from Samba. The user account gets locked. After unlocking the account, everything is back to normal. This happens several times a day to different users and from different Samba servers. All Samba servers have the same [global] configuration. Is this a Samba problem? Is samba perhaps not passing along the correct password? If I add

It was kind of working for a while but I could not hear the stream but it was running. I started messing with alsamixer and now it crashes. I see the error but I don't know how to fix it. ices-oss-xml: (alsa is the same just a different input module) <?xml version="1.0"?> <ices> <!-- run in background --> <background>0</background>

Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently

On Mon, Apr 20, 2015 at 06:21:22PM +0200, David Disseldorp wrote: > Hi Matthew, > > On Fri, 17 Apr 2015 13:11:53 -0400, Matthew Daubenspeck wrote: > > ... > > There are currently 113 printers installed, and those previous printers > > work as expected with no issues. Just suddenly anything added to cups > > cannot be installed. Any suggestions? I turned up the

Hi, I'm trying to diagnose a particularly elusive problem, and am wondering if anyone else here has seen anything similar and can offer any ideas. I have a conference bridge running Asterisk 1.2.32 (with slight mods), in a colo talking via a LAN to an ITSP using SIP/RTP. It is dedicated to a single customer. On several occasions over the last few months, the customer has reported instances

Hello, We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming calls from our carrier. The sip.conf looks like this: [kamailio1] type=friend host=10.0.0.1 context=incoming disallow=all allow=alaw All calls hit the incoming extension. In the extensions.conf we have multiple extensions configured, but now I have to add one which uses the special h extension to perform a CURL

Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

I am at a loss, I can terminate and receive calls via any of my providers with both IAX and SIP. I use GSM, G729a, and ulaw for those carriers. If I make an extension to extension call - there is no audio at all in either direction. All my extensions are set to use G729a (I have tried changing that though to see if it would fix it). I am fairly sure it is not a transcoding issue - as the

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have

Hello, Does anyone know how frequent segment faults occur in the current LTS release (version 11) and in the future LTS release (version 13)? We are currently using 1.6, which frequently throws unexplained segment faults, that's why we are considering to upgrade to the latest LTS version. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have a problem with Asterisk dropping audio. While in call, audio gets dropped for a while (from 5 to 60 secs, and obviously users often hangup, this means that I'm not sure the audio is always coming back after 60 secs), in the meantime the call remains up and no SIP signalation is generated. It happens randomly so it's very difficult to debug. I cannot see common

From: Doug <Doug at NaTel.net> >"Net Neutrality" is great in principle. But ISP's need to >somehow control those few percentage of users who suck down >a huge majority of the bandwidth. It's dollars and cents. There is a rational solution for the traffic management issue. It just needs to be aligned with the pricing model. I would like to see a tiered

At 11:08 AM -0500 7/12/04, Steven Critchfield wrote: >[snip] > >Would it be something people would like to be able to add filters to a >line? Consider normalization as a filter. Monitor could then be moved to >a filter as well. Echo cancel could be a filter. Set it up so multiple >filters could be added and chained together. This could help those with >echo chain a couple of

I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical

I'm using Samba 4.2.0 and cups 2.0.2. I have had the same print server and config since the early 3.x days and have never had an issue. Now suddenly, any new printers added to cups can no longer be installed in Windows. XP clients give a generic "Windows cannot connect to the printer. Either the printer name was typed incorrectly, or the specified priner has lost its connection to the

All I've suddenly lost incoming audio on my FWD connection. It worked fine up until Wed when all of the sudden my calls would complete but I couldn't hear any audio (I could see the status of the call on the CLI and could see that my call was using bandwidth on the ethernet switch and router). I swear I didn't change any of the configuration or even restart *, but all the sudden