similar to: Grandstream HT 503, asterisk 1.8 and TLS

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hey there i'm trying to get an Asterisk 11.11 with encryption working with my Grandstream phones. But i stuck. To avoid NAT problems i'm using IPv6 Just with TCP/TLS it's working fine. Only the SRTP funktion is not working. Asterisk tells me WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fa10800f5a0 (len 681) to [2a02:1205::...]:37635 returned -2: Success and also SSL

Hi all, It seems that the latest ATAs from Cisco/Linksys support SRTP. Did anybody give these features a go with asterisk? Regards Rajil

I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the "DMZ Host" in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 ====== NAT FW ====== * Server

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

Hi All, I am using "asterisk-11.12.0" version and I am trying to setup secure call (TLS + SRTP) between two extensions and while making a call, I got following error *CLI> == Using SIP RTP CoS mark 5 -- Executing [6004 at from-office:1] Dial("SIP/6003-00000000", "SIP/6004,20") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/6004 SSL certificate

Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce,

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Hi List, Just received a HT-286 Analog Telephone Adapter. This device is allows the user to take a standard analog telephone set and connect it to a VoIP/SIP based gateway. A low cost way of having PSTN devices make use of VoIP/SIP services. For example you could take a credit card processing machine and punch it into a location where you have broad band access. You could take a FAX

Just to tease people I'll be post a URL to my review of the HT-286 by thursday..

Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

On 1/23/2020 6:04 PM, hw wrote: >> This is what mine looks like which works just fine: >> >> [transport-tls] >> type = transport >> protocol = tls >> method = tlsv1_2 >> cipher = >> ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES128 >>

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

I am trying to backup a gmail account (not the one I am writing from) to dovecot, using doveadm-backup and imapc, but am having ssl connection problems. ted at expectation:~# doveadm backup -D -R -u ted imapc: dsync(ted): Info: imapc(imap.gmail.com:993): Connected to 74.125.71.108:993 (local 10.7.1.179:53852) dsync(ted): Warning: imapc(imap.gmail.com:993): Server disconnected unexpectedly:

> On 30/03/2020 22:11 Ben Mulvihill <ben.mulvihill at gmail.com> wrote: > > > I am trying to backup a gmail account (not the one I am writing from) > to dovecot, using doveadm-backup and imapc, but am having ssl > connection problems. > > ted at expectation:~# doveadm backup -D -R -u ted imapc: > dsync(ted): Info: imapc(imap.gmail.com:993): Connected to >

Hi, after switching from chan_sip to chan_pjsip, a device running Grandstream Wave leads to the following error message on the asterisk console: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> <SSL routines- ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 Something with the encryption must have changed with asterisk. How can I get the device to

Hello there, I just wasted some time setting up a Grandstream HT-488 to be used with Asterisk, so I thought I'd share the experience by writing a small tutorial at: http://astrecipes.net/index.php?n=338 Most tutorials I came across were for old versions of the firmware, and I spent too much time trying to figure out what the Forward to VoIP option wanted.... :-( I hope this helps,

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using