similar to: IAX2 dropped calls: need debug suggestions

Hi, I'm trying to track down why I'm getting calls dropped on an infrequent basis between two asterisk servers which are at the same physical location and connected to each other with UTP ethernet. Here is the connection diagram Asterisk Server 1 ===UTPENET== Switch ====UTPENET==== Asterisk Server 2 I see sporadic RX and TX frame retries when I enable iax2 debugging on either box.

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

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

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 Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Server B can place calls to Server A but when trying to place calls from Server A to Server B this is what I am getting:

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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 created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

Hello everybody, This is a problem disturb me for long. I run asterisk Asterisk 1.4.14 and A2Billing 1.3 in the same Debian 4.0 ETCH server. And there is also FoneBridge for TDM over Ethernet with E1 to make as well as receive calls from mobiles or PSTN. And IAX2 trunk runs between Asterisk and A2Billing. For most of the time they really do a good job. But some hours or days later----I mean it

Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when

All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EEXXXX where 'EE' is the exchange number and 'XXXX' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public

Hello list, I have this problem with dropped calls on Asterisk. The setup is SIP internal extensions (Grandstream GXP-2000), two internal analogue DAHDI extensions and IAX2 trunk lines. IAX2 trunks use ulaw/alaw. The Internet connection is ADSL. Asterisk is 1.6.1.6 Everything worked fine until about 1.5 months ago (for 1 year) until the client started to report dropped call. The scenario

Hi all, I'm experincing IAX2 call drops for about 20% of calls. I tried 'notransfer=yes' and 'jitterbuffer=yes' but to fail. My system configuration is like this. PSTN<========>Asterisk(TDM/Fxo 4port*3)<=====LAN(IAX2)=====>Iaxclient library And iax.con is... ----------------------------------------------- [general] port=5036 disallow=all allow=gsm

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be

Setup as follows: [private*] - Natting Router - [public*] [private*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest (Unspecified) (D)

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

What the heck does this mean? This is the first time I've seen this. Calls were going through ok for a couple weeks now. Dec 6 09:22:24 WARNING[1121866688]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/1400-45fb(4) to IAX2/simple/2(256) Dec 6 09:22:24 WARNING[1121866688]: app_dial.c:998 dial_exec: Had to drop call because I couldn't make SIP/1400-45fb

Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 00003ms SCall: 00130 DCall: 00000 [64.229.229.111:64823]* * USERNAME : 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: