similar to: Restricting registration for peer 'iaxmodem0' to 60 seconds

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

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

Hi, we have a faxserver with Asterisk, IAXModem and Hylafax. Faxes come from a SIP trunk to Asterisk, then are forwarded throught 5 IAXModems managed with Hylafax. Hylafax users can also send faxes to these modems and Asterisk send them throught the SIP trunk. We also have a dedicated modem used only for sending faxes coming from an Hylafax dedicated user. Sometimes Hylafax reports that a modem

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my

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

Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips. Next up, modem pool -- I wanted to know if the below config looked anywhere near half-sane for defining in asterisk what is essentially a small pool of four waiting modems that will handle faxes if another modem is busy: exten => _X.,1,Dial(IAX2/iaxmodem0/${EXTEN}) exten => _X.,2,Busy exten => _X.,3,Hangup

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

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

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

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

I'm running nut 2.0.1 with an MGE pulsar evolution 3000 with the mge-utalk (serial) driver on NetBSD 3.1 . I had a very strange occurance in my basement datacenter this early AM. the following was logged by upslog: 20080219 023452 100.0 123.0 049.0 [OL CHRG] NA 60.00 20080219 023522 100.0 122.8 049.0 [OL CHRG] NA 60.00 20080219 023552 098.0 123.0 049.0 [OL DISCHRG] NA 60.00 20080219 023622

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

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,