similar to: IAX2 call dropped when a second call comes in

Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04]

Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip extension / sip softphone (linphone) 5. 1 x 800Mhz Asterisk + Linux server 6. Asterisk version is 1.6.2.13 7. 1 x IAX2 incoming trunk from phone provider for 1

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

Hi, I'm trying to determine why IAX2 calls are getting dropped after a 4-24 hours of continuous connect time. My project requires that calls stay up for days at a time. When I turn on IAX2 debugging, I see "max retries exceeded" for control frames just before the connection is dropped. My test setup is: Zap Phone => Local Asterisk Server/IAX2/GSM => NAT => Internet =>

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

Hello, Just in case someone hasn't upgraded yet, and is using IAX2. -------- Original Message -------- Subject: AST-2009-006: IAX2 Call Number Resource Exhaustion Date: Thu, 03 Sep 2009 17:47:35 -0500 From: Asterisk Security Team <security at asterisk.org> To: bugtraq at securityfocus.com Asterisk Project Security Advisory - AST-2009-006

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

I have an analogue trunk to an AT&T Definity. It has a DISA context defined. From a Definity handset call the analogue port extension 1008 and wait for dial tone from asterisk. It takes between 3&4 rings. Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes nearly 10 secs to ring. Is this configurable? Ian Cowley -----Original Message----- From:

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

Hi people, i have problem with IAX2 between two asterisk PBX. When i try call some number i get "INVAL" packet, but when i try call same number via OpenVPN (is between this two asterisk) call is working fine.So i debug communications and here is my opinion ... Schema of connection: Asterisk1 -> ADSL router with NAT -> INTERNET -> Asterisk2 A)Calling directly via public

Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate ******** to XXX.XXX.XXX.XXX Using iax.conf it perfectly works. Maybe a bug in Realtime? TIA, Alex

I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes with T100P cards connected via IAX2. Calls between them work fine, but when I press a key at one end, it comes out the other end as a click, with no tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing call, and it sounds fine in that case; it seems to be only IAX2 that has the problem.

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols