similar to: No voice when the calls come from Internet

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? Thanks --------------

Hi All; I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were working fine via SIP, IAX and Digium fxo and fxs ports. Suddenly just before 2 or 3 days, the voice become garbage like robot when I place a call from the SIP Phone (which is in a country and the Asterisk box in another country). I am surprise what is the reason that let rtp become like this ! The sound now

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do

I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] ..... register => 688426:xxxxxxxx@fwd.pulver.com/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not

Hi All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is

A mi también me funciona para los dos casos: > dat <- read.csv("d11-16.csv", header=FALSE, sep=",", dec=".", skip=11, nrows=1) > dat V1 V2 V3 V4 V5 V6 V7 V8 V9 V10 V11 1 masa total en µg 30.04633 ug PEAKS MUY PEQUENOS NA NA NA NA NA NA NA > dat18 <- read.csv("d11-18.csv", header=FALSE,