similar to: Can't recieve Fax: No carrier detected - Ast erisk + iaxmodem + Hylafaxv --- sorry.wrong log.

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk and

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

When I have HylaFAX answer a call redirected to the fax extension in Asterisk when it detects CNG, Asterisk hangs up: Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 is ringing Jan 30 14:32:59 VERBOSE[1098]: -- IAX2/ttyIAX0552/12 answered Zap/23-1 Jan 30 14:32:59 DEBUG[1098]: Ooh, voice format changed to 8 Jan 30 14:33:01 VERBOSE[1098]: -- Channel 0/23, span 1 got hangup Jan 30

I have all three running on the same box. I say OT because it appears asterisk is doing it's job just fine. It must be an IAXmodem or faxgetty (hylafax) problem When faxes work, they look great. I have ten IAXmodems setup with different ports and they register fine. I have ten faxgettys that startup fine. I start the IAXmodems and then faxgettys in inittab. They are setup as a roll

I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk

Hello everyone, One more little problem with a %100 g729 setup. Native moh: musiconhold.conf: [default] mode=files directory=/mnt/kd/moh/default random=yes ; Play the files in a random order ls /mnt/kd/moh/default fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729 fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw Place a call on hold: Jun 1 14:55:30

Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

Hi, I've been playing round with Asterisk on Redhat 9 (2.4 Kernel) and was experiencing bad echo problems using Budgetone 100's when calling analogue lines in uk (Isdn4Linux / Digi Datafire). Calls to other ISDN and mobile network seemed ok although not much testing done. I've tried installing Asterisk on a faster processor (P4 3.0 GHZ) with a 2.6.8-24 kernel to see if that helps.

I think I'm very close to getting IAXModem and Hylafax going, but my current inbound hylafax logs show this: Feb 23 10:09:37.98: [ 3638]: MODEM <Empty line> Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier Two questions - 1) Does anyone know what step I missed here? (I.e. please help!) 2) Is there a document I should be working off of? Google doesn't seem to

I just upgraded a very old Asterisk installation to the last 1.2.31 I can find in Asterisk.org site. Now for some reason my IAX clients cannot connect to the server. I can do a "iax2 show peer iaxmodem1" and I get this: * Name : iaxmodem1 Secret : <Set> Context : oficina Mailbox : Dynamic : Yes Callerid : "" <> Expire

Hi, I've tired to forward a Fax from Asterisk to Hylafax. It works so far until I tried with a Fax machine. I just got error shown in the log below. I'm not sure why. I've tested it with other 6 machines and they all work fine. Do you have any idea why? Pim Hylafax Session log: Apr 12 11:16:48.82: [ 5933]: SESSION BEGIN 000000078 492212601860 Apr 12 11:16:48.82: [ 5933]:

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000

I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hello list , I?d like to announce possible problems with migrating any version prior to 1.0.2 to 1.0.3. Pay attention : 1. Codecs Codecs names/description have been changed . For example : versions <= 1.0.2 voip*CLI> show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. 1 (1 << 0)

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please