similar to: No audio one way

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

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, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though

Please tell me the obvious mistake I'm making here. (And yes, I well know about NAT and one-way audio problems in general.) I want to try the new T.38 passthrough stuff, downloaded it, built it, tested it with an SPA-2100 and can hear announcements fine but echo test shows no audio outbound (i.e. SPA to Asterisk). Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in

Hi All, Simple scenario: 7940 SIP><NAT Router><INTERNET><Asterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP Proxy><NAT Router><INTERNET><Asterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP

Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si,

I know when you read that subject everyone thinks NAT, but that isn't the case here. Incoming calls get 2 way audio, but outbound calls do not have incoming audio. below is the flow callee --> asterisk --> firewall/router --> provider Callee is firewalled, but not NAT. callee is on the same subnet as the asterisk box. Asterisk box has been completely excluded from the

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

I'm having the same "random" problem. I have "canreinvite=no" on all extensions. I have "qualify => yes" on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

Rich, thank you for your informative reply. I checked with our admin and he replied: "I setup from the start "nat=yes" and "canreinvite=no" on sip phones from Internet and modified the rtp channels (voice ports) and the rtp port on the phones. Still have the same problem, no sound." Perhaps the VPN solution is something we should try but this is more limiting than

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site),

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as

This email is intended to document an issue for anyone searching the archives. We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable conversation could be established due to extremely choppy audio in one direction only (outbound from * to distant sip phones and distant * boxes). We were running HEAD from June 8th. While diagnosing the root cause, we monitored bandwidth

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest