similar to: Getting T.38 issue

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400

Hi all the people in the list! I'm new on this list, this is my first post. I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with t38modem conected to hylafax as a sip extension of asterisk. Everything is supposed to be configured fine, the faxes start sending, but at the middle of the transaction, it fails. The T.38 termination provider told me that they were receiving

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid

Boa Tarde Lista. I'm having problems in tramiss?o a fax using T.38. My scenario is: Asterisk 1.6.0.5 2 ATA of Intelbras 2210. ReceiveFAX in the asterisk. Unable to fax when it is a ATA to another user on the Asterisk means, if I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX of Asterisk works perfect, but if I try to pass between two Branches

Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =

Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

I made a filesystem (mke2fs -j) on a logical volume under kernel 2.6.20 on a 64-bit based system, and when I try to mount it, ext3 complains with EXT3-fs: dm-1: couldn't mount because of unsupported optional features (80). I first thought I just forgot to make the filesystem, so I remade it and the error is still present. I ran fsck on this freshly made filesystem, and it completed with

Thanks for sharing the link. Actually I'm looking for a different approach without installing/using third party i.e. a user sends an email to Asterisk (which is also running mail service), as Asterisk receives the mail where the mail contains attachment and subject contains destination number, Asterisk will download the file and capture the number and later send fax to destination number just

[root at abc asterisk]# lsof -u 50771 | wc -l 0 BTW, I'm using CentOS 6.5 > > Date: Thu, 13 Oct 2016 10:20:19 -0400 >> From: Dovid Bender <dovid at telecurve.com> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users at lists.digium.com> >> Subject: Re: [asterisk-users] Openfile Issue >> Message-ID:

I did radius client status testing with radius server, able to access the radius server. However, still getting radius CDR issue after setting debug level 8 even granting 666 access to radiusclient-ng config files. message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded! Please advise if I missed out anything. Date: Mon, 26 Sep 2016 12:09:34 +0200 > From:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been seeing some strange issues recently as it relates to our Asterisk phone system. Let me give a little background on things: We have a T1 provided by Knology. We recently had an outage of Internet and phones. The following problems (except for the log messages) seemed to have cropped up after the outage. I was prompted by a friend of mine

Hi, Currently I'm experiencing crashes on Asterisk more recently, see messages below (crashed reason: segfault signal 6). abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT - dumping core asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fbd2c 00d170 (0) After running the backtrace for the

Hi Andrew and Willy, Thanks for sharing the info. As for enabling radius server debugging 'radiusd -X', made some test calls don't see the radiusclient sending data to radius server. However, using radtest or radiusclient testing, able to send data to radius server (after enabling debug). For further testing, on my other server using OpenSIPs, setup the radiusclient and data was

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13