search for: iax2

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 NoOp("SIP/6202-d193", "") in new stack -- Executing GotoIf(&quo...

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/pbxnet...

...PRI trunks to the IAX trunk with a CHANUNAVAIL error, whilst the SIP phones are actively calling out on those same PRI trunks. The numbers dialed are 10 digits with no prefix. I am hopeful that someone will recognize the issue and give me a pointer on where to look for the problem. - Registered IAX2 '4414' (AUTHENTICATED) at 192.168.1.102:4569 -- Accepting AUTHENTICATED call from 192.168.1.102: > requested format = alaw, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executin...

I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called

...ceeding passing it to SIP/test0-5821 -- Accepting UNAUTHENTICATED call from 195.66.73.122: > requested format = alaw, > requested prefs = (alaw|gsm), > actual format = alaw, > host prefs = (alaw|gsm), > priority = mine -- Executing Set("IAX2/soekris2-1", "CALLERID(number)=0327557574") in new stack -- Executing Dial("IAX2/soekris2-1", "Zap/1/327557671||tr") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/327557671 -- Zap/1-1 is proceeding passing it to IAX2/soek...

...d the format and number of digits sent. -- Accepting AUTHENTICATED call from 192.168.1.164: > requested format = alaw, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing Set("IAX2/4427-1", "EMERGENCYROUTE=YES") in new stack -- Executing Macro("IAX2/4427-1", "dialout-trunk|1|6167XXXXX||") in new stack -- Executing GotoIf("IAX2/4427-1", "1?3:2") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing M...

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 Firefly 1.9.8 build 3945....

Hey All, I am setting up a network of Asterisk servers using IAX2. I am wondering if it is possible to disable the handoff feature? At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is centrally hosted in a data centre. In addition the central machine has an IAX2 link to a VOIP provider (and might be set up with more in the future). All c...

Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the voice mail server where I have an IVR set to answer/handle the calls: [ivr-1] include => heading-out exten => h,1,Hangup exten => s,1,Set(LOO...

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 'IAX2/my-iax-provider-5647' ... and nothing happend.... the...

Hello all. I am trying to use app_mysql. It works for selects and functions, but does not want to work with procedures. Pls have a look: Calling function: CREATE FUNCTION f_1(a VARCHAR(20)) RETURNS INTEGER RETURN (SELECT count(*) from peer where name = a); Result: -- Executing Macro("IAX2/100-3", "local|100") in new stack -- Executing MYSQL("IAX2/100-3", "Connect connid localhost asterisk asterisk pbx_manager_new") in new stack -- Executing Set("IAX2/100-3", "QUERY=SELECT f_1('100')") in new stack -- Executin...

Hi, I have some questions regarding IAX, IAX2 and encrypted authentication. How can I know if IAX or IAX2 is used between two * servers? There is any guide about how to configure encrypted authentication (not in clear text)between two * servers? I "hear" on this list a couple of days ago that port 5036 is the default one for IAX an...

...takes the message. Just as expected. Now on, the Teliax side. Call comes in on Teliax trunk as 419 615 4152 as an IAX trunk. Call is switched down IAX trunk to Ottawa and phone rings at Ottawa. However, when call is sent to voicemail via Ottawa to vmail IAX trunk, caller gets slow busy back. IAX2 debug on vmail shows the call come in to 2502@default (just like PRI), it begins playout of call then immediateley hangs up. These are traces on the VM side. This is a trace of vm working (PRI - 419 993 2502) Verbosity was 0 and is now 9 -- Accepting AUTHENTICATED call from 10.149.32.11:...

...he call is making progress and puts through the incoming RTP I suppose. With an external anonymous ATA from the VOIP provider all works normally. Here is the debug: -- Accepted AUTHENTICATED TBD call from 192.168.19.130 -- Accepting DIAL from 192.168.19.130, formats = 0x4 -- Executing Macro("IAX2/200@200/2";, "dialout-trunk|2|0639006374|") in new stack -- Executing GotoIf("IAX2/200@200/2";, "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("IAX2/200@200/2";, "record-enable|200|OUT") in new stack -- Executing G...

...risk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels: *CLI> iax2 show channels Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format (None) 111.111.111.111 xxxxxx 00003/00020 00001/00003 00070ms 0001ms 0071ms g729 (None) 111.111....

I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/5555 or IAX2/8888) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 ext...

...erver_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set status='QUEUE',lead_id='0',uniqueid=' 1139506471.2', channel='IAX2/u32218094-1', callerid='unknown' where status = 'READY' and server_ip='127.0.0.1' and campaign_id='' and last_update_time > '19700101075955' limit 1;| |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0....

I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called xxxxx:xxxxxxx@iax2.fwdnet.net/18007425877 -- Call accepted by 65.39.205.121 (format ULA...

Dear List I have notice this kind of problem between my two * box. My scenario is : Iax GSM IaxClient----->PBX1------------>PBX2-->TDM today CVS Stable V1 I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join the two call i can see the log below from my PBX1, i can speak for few second and after the FireFly hangup. I have try to change * version from Stable to today CVS but no succ...

...cro(dialout-trunk,1,${EXTEN:1},) exten => _9.,2,Macro(dialout-trunk,2,${EXTEN:1},) exten => _9.,3,Macro(dialout-trunk,3,${EXTEN:1},) exten => _9.,4,Macro(outisbusy) ; No available circuits -----------------------------Sip Debug---------------------------- -- Executing GotoIf("IAX2/20@20/4", "1?5:8") in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget("IAX2/20@20/4", "RecEnable=RECORD-OUT/20") in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=20 -- DBget: Value not found in database. -- Execut...