similar to: SIP disconnecting : response 481

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

regarding my previous post about SIP outgoing calls dropping with an error 481 .. this is my output from a SIP debug. the call dropped occurs at the end. Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my control. help :) please!! Dave Signal=0 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP

Hi, I am using SIP extensions connected to the PSTN with the CAPI Channel driver. All works fine except that one of the sip phones keeps ringing when the caller hangs up before extension is answered. The phones are grandstream 100, though we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

Hi all: I've no response for the last question with the same subject. Please excuse me for the extreme length of this mail, but I send 2 SIP traces. I have problem with * and 5300, when the incoming and outgoing call are routed thru the same SIP gateway (AS5300). Do I need to set an special things in sip.conf? First all, the * printout. Second, the 5300 trace. Thanks in advace, Gus

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

People: Did you have seen this message before? noc2pbx*CLI> -- Executing Goto("SIP/-081363a0", "ivr1|2000|1") in new stack -- Goto (ivr1,2000,1) -- Executing Answer("SIP/-081363a0", "") in new stack -- Executing Wait("SIP/-081363a0", "1") in new stack -- Executing BackGround("SIP/-081363a0",

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

Hello: New to asterisk and hoping to use for http://summitcamp.org research station. While trying to use with Inphonex I find that incoming calls drop after about one minute-- -- Got SIP response 420 "Bad Extension" back from 208.239.76.169 == Spawn extension (incoming-inphonex, 210, 1) exited non-zero on 'SIP/inphonex-095bf208' Found that I can use `*CLI> sip

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not