similar to: Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax

Hi all, after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show what's happens. I've trying setting debug and verbose to 100 but nothing, no show. All commands works as expected but i can't what's happens on my asterisk server. asterisk*CLI> core show settings PBX Core settings ----------------- Version: 14.2.0 Build Options:

Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages:

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

I can't seem to change the default registration for IAX clients: Feb 6 12:22:52 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'virbiage' to 60 seconds (requested 3600) Feb 6 12:23:03 NOTICE[7883]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'test1' to 60 seconds (requested 1200) Can this be controlled on a

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel

I have setup hylafax today, along with iaxmodem. I'm just starting the debugging process and see the following message every 60 seconds: [Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry: Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300) Can someone tell me what this means? Why is it there? And how do I get rid of it! Thanks, MD

I have never noticed the message prior my upgrade of CVS head: chan_iax2.c:5589 update_registry: Restricting registration for peer '611' to 60 seconds (requested 1200) What does it mean, and how can I fix it? 611 is a firefly soft phone. bye Ronald Wiplinger

Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [fax at phones:1] Set("Zap/2-1", "FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif") in new stack -- Executing [fax at phones:2] RxFAX("Zap/2-1", "/var/spool/asterisk-fax/1207322398.0.tif") in new st ack [Apr 4

Dear Sirs, we currently manufacture/import UPS units in Italy and distribute them all over the country. How can we obtain an official certification to offer our products with full compliance with QNAP, Synology and similar hardware? Looking forward to hearing from you, Best Regards ________________________ Alessandro Gualtieri LAER Accumulatori Srl Via Pio XII, 65/c 20060 - Pozzuolo

The problem with using Voice Modems is that they fall into two categories: 1) Hardware Modems which only have half-duplex transmission of voice 2) Soft/Win/Lin modems which are proprietry and don't have asterisk drivers Please shoot down this recipe before I waste any time trying to acheive it: Rationalisation: ---------------- In Australia and I assume other places, there is no

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

hi, I''ve a dedicated internet link 512k with a cisco router, and now, I want to upgrade bandwidth with 8 ADSLs (512k each) then: - build kernel 2.4.27 with pachs: - patch-o-matic-ng-20040621 : CONNMARK, H323, etc... - http://www.ssi.bg/~ja/#multigw Julian''s patch. - Activate "equal cost multi path" (Advanced Router-Networking Options) and so on -

Hello everybody, we're trying to set-up Samba to share directories with Win users from some Linux servers. We've set up kerberos, gotten a ticket, joined the server to the domain, we get correct users/groups from "wbinfo" and "getent". The problem lies in "id": it does not update its user<->group mappings when they change on AD, even if

There's a #define macro in channels/chan_iax.c that you can modify to make this forced value higher. Just open it up in your favourite editor and search for '60' and you'll find it. Now if there's an easier way than having to change a source-level macro, I'm all ears... Cheers!, --jkinard -----Original Message----- From: asterisk-users-bounces at lists.digium.com

Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten => 999,1,Dial(Zap/g1/02601591) exten => 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 I am able to

Hi, i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think. marcus -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Gesendet: Donnerstag, 6. April 2006 11:50 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Dial out on Zap Hi,

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

Hi all iam using Iptables+TC+HTB on Redhat 9 working fine with the bandwidth control I am taging my eth1 with Vlan interface with Cisco Switch when even though i have mention ceil, its crossing more than Ceil, when they are effected Virus in their network or DoS attacks, its should be control the mentioned Ceil right, why this uploads are increaing.. when the uploads are increased all the

Hello, I have anolog lines coming throug Dahdi to Asterisk Server, one of the anolog lines is used for fax line. I received fax fine without any problems using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax