Re: [asterisk-users] One way audio on outgoing calls

2020-08-07 Thread Administrator
Hi Carlos Le 07/08/2020 à 06:33, Carlos Chavez a écrit :     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, o

[asterisk-users] One way audio on outgoing calls

2020-08-06 Thread Carlos Chavez
    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 anyth

Re: [asterisk-users] One way audio on new build

2020-02-25 Thread Joshua C. Colp
On Mon, Feb 24, 2020 at 10:59 PM Ira wrote: > Hello Asterisk, > > I've been running a CENTOS 5 box with Asterisk 14 and am trying to > move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk > from Source as I've always done and copied all the configuration files > and other stuff f

[asterisk-users] One way audio on new build

2020-02-24 Thread Ira
Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it a

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

2015-08-15 Thread D'Arcy J.M. Cain
On Sat, 15 Aug 2015 12:42:38 -0300 Joshua Colp wrote: > > I am not sure why this hasn't bit anyone else. Perhaps most > > Asterisk systems are in one of two classes, connecting to all NAT > > phones or connecting to all public phones, and I am in a minority > > situation where I am talking to a m

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

2015-08-15 Thread Joshua Colp
On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote: > On Sat, 15 Aug 2015 16:30:39 +0800 > Michael Dupree wrote: > > Not 100% ure, but maybe play with the canreinvite or directmedia > > settings. > > Yes! That was it. Just for future searches here is what I did. I > added "directmedia =

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

2015-08-15 Thread D'Arcy J.M. Cain
On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree wrote: > Not 100% ure, but maybe play with the canreinvite or directmedia > settings. Yes! That was it. Just for future searches here is what I did. I added "directmedia = no" in sip.conf. This fixed the issue. I believe that Asterisk was get

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-15 Thread Michael Dupree
Not 100% ure, but maybe play with the canreinvite or directmedia settings. On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain wrote: > I have been banging my head against the wall for weeks now on this > one. I have a switch running NetBSD and Asterisk 11.19.0 although I > have had this problem

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-13 Thread Stefan Viljoen
Hi D'Arcy >> that the server IP for RTP as specified in the initial SIP is correct? >Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. This was a problem we had when the RTP server negotiated in SIP with our VOIP ITSP on one side of t

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-13 Thread D'Arcy J.M. Cain
On Thu, 13 Aug 2015 10:41:31 +0200 "Stefan Viljoen" wrote: > Have you checked your RTP port ranges (I'm sure you have), and also Yes. The ATA is using a range well within the range open on the server. > that the server IP for RTP as specified in the initial SIP is correct? Both the server and

[asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-13 Thread Stefan Viljoen
To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue Message-ID: <20150811151044.79872ce9@imp> Content-Type: text/plain; charset=US-ASCII Given that both of us can make and accept call

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-12 Thread Joshua Colp
On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain wrote: > I have been banging my head against the wall for weeks now on this > one. I have a switch running NetBSD and Asterisk 11.19.0 although I > have had this problem on older versions as well. I, and my users, can > call out, we can receive

[asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-11 Thread D'Arcy J.M. Cain
I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The d

Re: [asterisk-users] One way audio internal

2014-11-21 Thread Andrew Colin
: Friday, November 21, 2014 1:04 PM To: Andrew Colin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] One way audio internal Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, "Andrew Colin" wrote: I am using the

Re: [asterisk-users] One way audio internal

2014-11-21 Thread A J Stiles
On Friday 21 Nov 2014, Andrew Colin wrote: > I am using the free g729 > OK, so there shouldn't be any licencing problems (unless for some reason your Asterisk is wanting to use the paid-for g.729 aot the Free one. Look at the CLI output very, very carefully to see if this might be happening).

Re: [asterisk-users] One way audio internal

2014-11-21 Thread Andrew Colin
I currently am running on a Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz Codec im using is codec_g729-ast18-icc-glibc-x86_64-core2.so -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Ast

Re: [asterisk-users] One way audio internal

2014-11-21 Thread Mitul Limbani
are not liable for any such corruption, interception, amendment, > tampering or viruses or any consequences thereof. > > > > *From:* Mitul Limbani [mailto:mi...@enterux.in] > *Sent:* Friday, November 21, 2014 12:51 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Cc:*

Re: [asterisk-users] One way audio internal

2014-11-21 Thread Andrew Colin
Subject: Re: [asterisk-users] One way audio internal You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, "A J Stiles" wrote: On Friday 21 Nov 2014, Andrew Colin wrote: > Hi All > > We have a strange issue with our hosted asterisk server running o

Re: [asterisk-users] One way audio internal

2014-11-21 Thread Mitul Limbani
You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, "A J Stiles" wrote: > On Friday 21 Nov 2014, Andrew Colin wrote: > > Hi All > > > > We have a strange issue with our hosted asterisk server running on Debian > > Internal calls btween extensions using g729 give one way

Re: [asterisk-users] One way audio internal

2014-11-21 Thread A J Stiles
On Friday 21 Nov 2014, Andrew Colin wrote: > Hi All > > We have a strange issue with our hosted asterisk server running on Debian > Internal calls btween extensions using g729 give one way audio > As soon as we change the codec to ALAW the issues goes away. > > Any ideas how to fix this? > > Out

[asterisk-users] One way audio internal

2014-11-21 Thread Andrew Colin
Hi All We have a strange issue with our hosted asterisk server running on Debian Internal calls btween extensions using g729 give one way audio As soon as we change the codec to ALAW the issues goes away. Any ideas how to fix this? Outbound calls via a trunk work fine with g729 Kind

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-22 Thread Gary Shergill
31 -> 192.168.3.150 UDP 224 Source port: 16514 Destination port: 65021 Thanks again for your time! Kind Regards, Gary Shergill - Original Message - From: "Amit Patkar" To: asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2014 4:55:57 PM Subject: Re: [asterisk

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread bhavik patel
hergill > > > - Original Message - > From: "Gary Shergill" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, May 21, 2014 3:36:54 PM > Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external &

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread Amit Patkar
thers, with no obvious reason either way. Thanks again. Kind Regards, Gary Shergill - Original Message - From: "Gary Shergill" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-user

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread Gary Shergill
ercial Discussion" Sent: Wednesday, May 21, 2014 3:36:54 PM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Amit, ICE/STUN is configured correctly. The extension for the webrtc user is defined in sip.conf on the asteriskrtc.local server. The other user i

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread Gary Shergill
um.com Sent: Wednesday, May 21, 2014 04:41:50 AM Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk) Hi Gary You need to check if ICE / STUN is configured. How are these extensions configured? If you are in private network, you might have to disable DirectMedia / reInv

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread Amit Patkar
Hi Gary You need to check if ICE / STUN is configured. How are these extensions configured? If you are in private network, you might have to disable DirectMedia / reInvite for calls going between 2 asterisk boxes. I hope this helps to resolve your issue. *Thanks & Regards,* Amit Patkar On 5

[asterisk-users] One Way Audio with WebRTC (with external asterisk)

2014-05-21 Thread Gary Shergill
Hi, I've run into a slight issue when using WebRTC and two Asterisk boxes. I am using SIPml as the test WebRTC client. My two asterisk boxes, one of them is configured for WebRTC with websockets, etc (asteriskrtc.local) and the other is just a standard asterisk server (asteriskgary.local). De

[asterisk-users] One-way audio with media_address

2012-09-04 Thread Richard Kenner
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small patch to allow specifying an address for RTP media. That worked. In 10.7.0, this appears to be built in with "media_address", but it doesn't work for me. My Asterisk server has multiple addresses, all global address on two di

[asterisk-users] One-way audio when calling multiple SIP

2012-06-24 Thread Roland
Hi, On one of our locations, I am having issues with one-way audio when I call several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the phones individually, they work fine, so it's not a volume setting on the phone. Also this setup has worked at other locations. Any idea's what to

Re: [asterisk-users] One way audio when using originate...

2011-08-13 Thread Pezhman Lali
Dear in normal mode, .call files make a call between the system and who you named remote person, I don't know where are you? in natmode=yes, set qualify=yes. check the negotiated codecs also. Best On Sat, Aug 13, 2011 at 1:29 AM, Carlos Chavez wrote: >We are having a problem when trying t

[asterisk-users] One way audio when using originate...

2011-08-12 Thread Carlos Chavez
We are having a problem when trying to use originate or AMI to make a call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to call the PSTN. When dialing from IP phones everything works fine. When you try making the call with originate, AMI or a call file then the remote pe

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Tim King
Still not working now that audio is restored jitter makes it inaudible? I am ready to move this to commercial if someone can tell me how I need to pay for support, Thanks Tim On Thu, Mar 10, 2011 at 10:19 AM, Tim King wrote: > It looks like the issue was my provider enforcing a codec translat

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Tim King
It looks like the issue was my provider enforcing a codec translation that was not working. On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel wrote: > Also it could be the routing issue as well. > > -- > Sent from my iPhone > > On Mar 9, 2011, at 7:43 PM, Duncan Turnbull wrote: > > So that suggests

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Satish Patel
Also it could be the routing issue as well. -- Sent from my iPhone On Mar 9, 2011, at 7:43 PM, Duncan Turnbull wrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 > 74.204.4.5.11732 IP 74.204.4.5.11732 > 209.216.2.203.60362 Somewhere n

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Tim King
My message with the configuration attached is awaiting moderator approval. I will try to paste relevant data here. *sip.conf* [general] context=inbound ; allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw dtmfmode =

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Steve Davies
On 10 March 2011 11:17, Ishfaq Malik wrote: > Just fixed our problem with > > directmedia=no > > but this only applies if your extensions are behind a nat > > Ish > There are several reasons why "directmedia=no" might be the correct configuration. 1) NAT - probably the most common reason 2) Rout

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Ishfaq Malik
Just fixed our problem with directmedia=no but this only applies if your extensions are behind a nat Ish On Thu, 2011-03-10 at 09:40 +, Ishfaq Malik wrote: > I've been having a similar (well exactly the same) problem this last > week and have been bashing my head trying to fix it. > > Just

Re: [asterisk-users] One Way Audio

2011-03-10 Thread Ishfaq Malik
I've been having a similar (well exactly the same) problem this last week and have been bashing my head trying to fix it. Just one question, are you using RealTime? Ish On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote: > I am having trouble with no return audio on inbound calls. I have been > w

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Jai Rangi
You can use this link too. http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale Keep the context as context=from-trunk. -Jai On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi wrote: > > 209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. > > BTW Did y

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Jai Rangi
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Duncan Turnbull
So that suggests audio is flowing as follows in a unidirectional manner > 199.173.66.22.53102 > 74.204.4.5.11732 IP 74.204.4.5.11732 > > 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box > 19:18:36.389548 IP 74.2

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Tim King
It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Duncan Turnbull
Can you do a tcpdump to look at the rtp streams on your box and check they are both generating and aiming at the right places IAX will have no issue with NAT/firewall but SIP / RTP can. tcpdump -nn udp and portrange 1-2 (pick your portrange if its operating on something else) Should s

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Tim King
Thank you I have also tried those settings. The main thing is coming from my voip provider all I am doing is bridging the calls to two other devices (1 trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with an IVR and when I call in I can not hear the IVR. However if I call dire

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Satish Patel
What about your sip clients? Are they on public network? Try on sip.conf Nat=no/yes conreinvite=yes/no -- Sent from my iPhone On Mar 9, 2011, at 6:11 PM, Tim King wrote: IPTBALES is off and I have all firewalls disabled. All network elements currently involved have public IP's assigned t

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Tim King
IPTBALES is off and I have all firewalls disabled. All network elements currently involved have public IP's assigned to them. My main asterisk box has a public IP. I have multiple trunks to voip peers for inbound and outbound calls which are also all public IP's. My two clients are trunked via IAX

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Adolphe Cher-Aime
How is your network is organized? Is your server behind a firewal, about iptables ? On Wed, Mar 9, 2011 at 5:40 PM, Tim King wrote: > I am having trouble with no return audio on inbound calls. I have been > working on this for 18 hours and even built a fresh server and moved > everything over

Re: [asterisk-users] One Way Audio

2011-03-09 Thread Paul Belanger
On 11-03-09 05:40 PM, Tim King wrote: > I am having trouble with no return audio on inbound calls. I have been > working on this for 18 hours and even built a fresh server and moved > everything over and I am getting the same results. I need someone that can > help get this resolved tonight. I can

[asterisk-users] One Way Audio

2011-03-09 Thread Tim King
I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Plea

[asterisk-users] One way audio problem

2010-11-17 Thread Deepika Nijhawan
Hi, Asterisk is making a call to a peer. In 200 ok, peer is sending its application server ip in contact field, so asterisk sends ACK to that IP. RTP starts flowing between endpoints and peer plays an IVR and asks for destination number. After entering destination number peer's application serv

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Flavio Miranda
FORWARD -p UDP --dport 5060 -j ACCEPT Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda > Date: Thu, 16 Sep 2010 18:45:38 -0400 > From: paul.belan...@polybeacon.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] one way aud

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Paul Belanger
On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson wrote: > The server is not behind NAT only the client above is > Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabbe

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Thomas Johnson
I already have that covered [tomfmason] type=friend secret=secret callerid="Thomas Johnson" host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip The server is not behind NAT only the client above is On Thu, Sep 16, 2010 at 4:59 PM, Paul Belang

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Paul Belanger
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson wrote: > Also, if I disable the firewall in my router I lose incoming audio and > outgoing audio works. > http://www.aocomputing.net/?p=3 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freeno

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Thomas Johnson
I have tried doing that with just ulaw and alaw, respectively, and nothing changed Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. On Thu, Sep 16, 2010 at 2:50 PM, Sebastian wrote: > > > On 09/16/2010 07:59 PM, Thomas Johnson wrote: > > the client

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Sebastian
On 09/16/2010 07:59 PM, Thomas Johnson wrote: > the client that is behind nat is > [tomfmason] > type=friend > secret=secret > callerid="Thomas Johnson" > host=dynamic > nat=yes > canreinvite=no > disallow=all > allow=gsm > allow=ulaw > allow=alaw > qualify=yes > context=sip > > do I have to ena

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Thomas Johnson
the client that is behind nat is [tomfmason] type=friend secret=secret callerid="Thomas Johnson" host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? On Thu, Sep 16, 2010 at 1:36 PM, Sebastian wrote: > >

Re: [asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Sebastian
On 09/16/2010 06:58 PM, Thomas Johnson wrote: > I am having a one way audio issue with xlite clients behind NAT. They > can connect to the server and make calls but no audio is heard on the > other end. > > my sip conf > > [general] > context=default > bindport=5060 > bindaddr=0.0.0.0 > srvlookup

[asterisk-users] one way audio for xlite clients behind NAT

2010-09-16 Thread Thomas Johnson
I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="T

Re: [asterisk-users] One way audio when overlapdial is set to yes

2010-09-15 Thread leonimar cape
t: [asterisk-users] One way audio when overlapdial is set to yes > > Hi Group, > > > I am currently facing a dead end and any help will be much appreciated. > > I have an a104d installed in an asterisk box, two of which is configured on >ISDN > > pri. One is

[asterisk-users] One way audio when overlapdial is set to yes

2010-09-15 Thread leonimar cape
Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath t

[asterisk-users] One way audio when dialing multiple registrations

2010-07-21 Thread Nasir Javaid
Hi again today when i was doing my research on this issue i found that even dialing a sip user by it's IP also raises this problem. here is what i did, First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I diale

[asterisk-users] One way audio when dialing multiple registrations

2010-07-20 Thread Nasir Javaid
sorry for the typo mistake. the actual dial string that I used is like this Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT) it is not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT) it was just a typing

Re: [asterisk-users] One way audio when dialing multiple registrations

2010-07-19 Thread Zeeshan Zakaria
I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one. Considering how SIP communication works, I believe SIP doesn't allow multiple registrations like this. Maybe somebody can correct me here if I am wrong. Zeeshan A Zakaria -- www.i

[asterisk-users] One way audio when dialing multiple registrations

2010-07-19 Thread Nasir Javaid
thanks a lot zishan and philipp, probably that is the problem that is occurring. I am gonna take some wireshark or etherial trace to further investigate the problem. i don't wanna stuck into port forwarding issue as it will waste lot of time and also normal calling is working on my current port fo

Re: [asterisk-users] One way audio when dialing multiple registrations

2010-07-19 Thread Zeeshan Zakaria
Hi Nasir, Please don't send me direct emails, unless you want to secure my paid consultancy services or want to do some other business. For setting up the RTP, you need to do it on your firewall, which is receiving RTP traffic from these particular IP address. I can't guess how to do it on your r

Re: [asterisk-users] One way audio when dialing multiple registrations

2010-07-16 Thread Zeeshan Zakaria
Based on the info you provided (though wireshark analysis will tell more about it), I am sure what is happening is that rtp coming back from the called doesn't know which ip to go to, because asterisk knows two ip addressses for the same extension due to the way you dialed it, i.e. in ringgroup fas

Re: [asterisk-users] One way audio when dialing multiple registrations

2010-07-15 Thread Philipp von Klitzing
Hi! > I am working on calling 2 registrations of same user on 2 different ip or > ports. It works fine and both phones ring simultaneously. the problem is > that there is one way audio, calling party can hear me but i can't hear > calling party. You need to make sure that these two phones use *di

Re: [asterisk-users] One way audio when dialing multiple registrations

2010-07-15 Thread Jonas Kellens
One-way audio is mostly firewall problem. Are you behind firewall ? You can check the audio-ports that are being used in the SDP-message by doing a /sip debug/. Maybe you do not have enough UDP-ports open for the audio ? Jonas. On 07/15/2010 04:38 PM, Nasir Javaid wrote: Hi, I am workin

[asterisk-users] One way audio when dialing multiple registrations

2010-07-15 Thread Nasir Javaid
Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060 SIP/x...

[asterisk-users] One way audio problem, a=sendonly and a re-invite

2010-05-12 Thread David Cunningham
Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing

Re: [asterisk-users] One-Way Audio after Hold

2010-02-17 Thread Mike Diehl
Brent Torrenga wrote: I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places th

[asterisk-users] One-Way Audio after Hold

2010-02-17 Thread Brent Torrenga
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold

[asterisk-users] One way audio with Grandstream HT503

2010-02-01 Thread jonas kellens
Hello list ! I'm having one way audio on incoming and outgoing calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DE

Re: [asterisk-users] One Way Audio from External Sip Soft & Hard Phone

2009-07-07 Thread Steve Totaro
On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgar wrote: > I have a problem with one way audio on Sip and I guess it may be a NAT > issue, in the example below 204 is rung by 208 (xlite external) > > > > I dial perfectly but when I get to the answering of the Asterisk, I can hear > audio from the Asterisk

[asterisk-users] One Way Audio from External Sip Soft & Hard Phone

2009-07-07 Thread Paul Edgar
I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voi

Re: [asterisk-users] One way AUDIO

2009-04-07 Thread Danny Nicholas
pen. Do a netstat -an during each call and see what is different. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Monday, April 06, 2009 6:04 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] One

Re: [asterisk-users] One way AUDIO

2009-04-06 Thread David @ULC
Few Running figures !! On Tue, Apr 7, 2009 at 3:41 AM, David @ULC wrote: > > I have a server with 2 Lan Cards. > > Now, when I am trying to make calls using Local Lan, its One way Audio > which means customer cant hear me but if I use Static IP with Wan > Connection, it works perfectly. > > I ch

Re: [asterisk-users] One way AUDIO

2009-04-06 Thread Giancarlo Rubio
How tcpdump on interface show?? 2009/4/6 David @ULC : > > Can it be that any Port got blocked ? > > On Tue, Apr 7, 2009 at 3:41 AM, David @ULC wrote: >> >> I have a server with 2 Lan Cards. >> >> Now, when I am trying to make calls using Local Lan, its One way Audio >> which means customer cant

Re: [asterisk-users] One way AUDIO

2009-04-06 Thread David @ULC
Can it be that any Port got blocked ? On Tue, Apr 7, 2009 at 3:41 AM, David @ULC wrote: > > I have a server with 2 Lan Cards. > > Now, when I am trying to make calls using Local Lan, its One way Audio > which means customer cant hear me but if I use Static IP with Wan > Connection, it works perf

[asterisk-users] One way AUDIO

2009-04-06 Thread David @ULC
I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no u

[asterisk-users] One way audio after IVR tree

2009-02-07 Thread James Lamanna
Hi, I have a couple of users who are having a peculiar problem. On some outbound numbers where there is a deep IVR tree (3+ selections), and then a live person picks up, the live person will be unable to hear them on the phone, but they can hear the live person. I've done packet traces and it appea

Re: [asterisk-users] One Way Audio Problem

2008-10-17 Thread Brent Davidson
GNUbie wrote: > What particular configs are you looking for? Below is my current setup > and scenario: > > [snom] ==LAN==> [asterisk] ==FXO/POTS ==> [analog_telephone/mobile_phone] > > SNOM is using the 192.168.101.102 IP address > Asterisk is using 192.168.101.1 IP address for its eth1 interface >

Re: [asterisk-users] One Way Audio Problem

2008-10-17 Thread Jeff LaCoursiere
On Thu, 16 Oct 2008, GNUbie wrote: > Hello, > > On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere <[EMAIL PROTECTED]> wrote: > > > > A packet trace will probably show exactly what is happening. Try: > > > > tcpdump -nlXs 8192 -i eth0 port 5060 > > > > You should be able to see the SIP informati

Re: [asterisk-users] One Way Audio Problem

2008-10-16 Thread Karsten Wemheuer
Hi, Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie: > Hello Karsten, > > On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote: > > > > Please post Your sip.conf. > > Which IP-Address do You configure in the snom for Your asterisk? (eth0 > > or eth1)? > > The SN

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Tzafrir Cohen
On Thu, Oct 16, 2008 at 09:22:01AM +0800, GNUbie wrote: > Hello Daniel, > > On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker > <[EMAIL PROTECTED]> wrote: > > Might be a stretch, but does the Asterisk log show that the call was > > answered? I had this problem when interfacing * with an NEC sys

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread GNUbie
Hello Steve, On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > canreinvite defaults to yes, whether specified or not. > > http://www.voip-info.org/wiki/view/tips > > If you follow these directions adapting to your particular > circumstances and it doesn't work, post your

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Steve Totaro
Sorry, wrong thread, time for bed. I thought this was the thread where the guy was having issues with one way audio on his third call, and his Asterisk server was behind NAT. Good night everyone and have pleasant dreams of 700 point drops in the DOW! OT, did you know if the government took the $

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Steve Totaro
Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie <[EMAIL PROTECTED]> wrote: > Hello Steve, > > On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro > <[EMAIL PROTECTED]> wrote: >> Did yo

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread GNUbie
Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Steve Totaro
canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewh

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Steve Totaro
Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie <[EMAIL PROTECTED]> wrote: > Hello Karsten, > > On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote: >> >> Please post Your sip.conf. >> Which IP-Address do You configure in the snom for Your asterisk

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread Steve Totaro
Did you try it the magic number of times, three? On Sun, Oct 12, 2008 at 9:57 PM, GNUbie <[EMAIL PROTECTED]> wrote: > Hello Tzafrir, > > On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: >> >> This means Zaptel gets silence from Asterisk. >> >> What codecs are used? What do

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread GNUbie
Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote: > > Please post Your sip.conf. > Which IP-Address do You configure in the snom for Your asterisk? (eth0 > or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. Th

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread GNUbie
Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote: > Might be a stretch, but does the Asterisk log show that the call was > answered? I had this problem when interfacing * with an NEC system to > do call parking pickup. The NEC would never give a dialton

Re: [asterisk-users] One Way Audio Problem

2008-10-15 Thread GNUbie
Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere <[EMAIL PROTECTED]> wrote: > > A packet trace will probably show exactly what is happening. Try: > > tcpdump -nlXs 8192 -i eth0 port 5060 > > You should be able to see the SIP information going back and forth and > will probably show you t

Re: [asterisk-users] One Way Audio Problem

2008-10-13 Thread Karsten Wemheuer
Hi, Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie: > Hello Gordon, > > On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson > <[EMAIL PROTECTED]> wrote: > > > > You mention the SIP phone being inside the LAN. Where is the Asterisk box? > > It is the main gateway of the IP phones and my lapt

Re: [asterisk-users] One Way Audio Problem

2008-10-13 Thread Daniel Hazelbaker
Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only

Re: [asterisk-users] One Way Audio Problem

2008-10-13 Thread Jeff LaCoursiere
A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewal

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