similar to: No audio for outgoing calls

Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

    I am having a strange problem with a new provider.  We already have a couple SIP trunks working fine.  We are trying a new provider but we are having one way audio problems with outgoing calls.  Incoming calls do have two way audio, only outgoing calls have this problem.  I do not see anything odd with a packet capture and using PJSIP history to check.  The provider says that on outgoing

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

That was exactly it! Thanks you VERY much! Mike ---- For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > > Hi, > > I've had a bit of a problem with one way audio, and it happens exactly when > I believe it shouldn't (and works perfectly when I would guess I could

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hi- let me start off by saying I'm a newbie to Asterisk and this list and I'll also apologize up front for stupid questions. I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel gateway set up. I used the configurations from the O'Reilly article and I haven't even set up voice mail (the only change was to add the iaxtel entry). My problem is the audio out from my

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

Over the past 2 weeks I have been able to compile and get an asterisk system up & running on a debian Linux box. I have setup 5 internal sip clients on the lan and all works great! I can also call from outside (PSTN) into the system and reach extensions and services no problem. All is up & running behind a nat firewall with proper ports forwarded and locked down on each device to work

Hello, I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied registration info and it does work. We can also register Asterisk with FWD and dial the FWD

Hi, everybody: I have just set up a system with a regional VOIP provider. I have two IAX channels to this provider. Incoming calls ring a configured SIP extension immediately, but outgoing calls are delayed for about 8 to 10 seconds before the remote PSTN end starts ringing: > -- Called [IAX2 channel] > -- Call accepted by [IAX2 provider IP] (format ulaw) > -- Format for

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

Dear list: When I call an specific number on the PSTN, the provider who holds the destination number give back an specified sound just after admitting their incoming calls. Is there a way to allow Asterisk to compare sounds received to decide what is the Telco answering the call? I'm planning to do it to select the right provider to route further calls at least cost. In my country there are

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

I have my Asterisk server all setup. But have an odd problem and hope someone here can help. I have a Polycom IP 300, a Grandstream GXP-2000, and an installation of X-Lite. They can each call each other just fine (extension-to-extension). I can also dial-in from the outside (via Broadvoice) and can leave and retrieve voicemails. When I set ANY of the extensions (clients mentioned above) to

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has

Hi, I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone. I have to increase the rxgain and txgain