similar to: PJSIP does not stop sending invites after call is canceled

PJISP cannot handle the From field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: <?>: sip_transport.c Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax error exception when parsing 'From' header on line 4 col 40: CANCEL sip:14408785990 at 162.255.138.102:5060 SIP/2.0 Via: SIP/2.0/UDP

I got the response below from a provider. How do I extract the Identity header and apply it to the next INVITE? Is it possible at all with PJSIP? SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 172.16.7.254:52169 ;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04 Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI From: "Peter Perez" <sip:727XXX3019

On 12/30/2017 08:18 PM, Dovid Bender wrote: > Script kiddies trying to find vulnerable systems that they can make > calls on. Lock down the box with iptables and use fail2ban to block > them. The via is probably bogus unless a box at the DoD was comprimised. > > > > On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057 From:

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.33.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.33.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

On 21/06/2019 15:39, Edouard Guign? via samba wrote: > Hello, > > I am facing 2 issues now. > The first one is the more critical for me... > > 1. When I switch from sssd to winbind with : > # authconfig --enablekrb5 --enablewinbind --enablewinbindauth > --enablemkhomedir --update > > My sftp access did not work. Does it change the way to pass the login ? > I used

Hi Steve, - Your right - the file was AMI (copied the other one). By direct connect I simply meant the speaker is an extension on that server. here is the SIP debug <--- SIP read from UDP:X.X.X.X:1024 ---> == Using SIP RTP CoS mark 5 Audio is at 16060 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably

Yes, I have only one domain. Even after added "winbind use default domain = yes" to smb.cnf, I cannot ssh : /Jun 21 12:43:59 [localhost] sshd[5938]: pam_sss(sshd:auth): Request to sssd failed. Connection refused// //Jun 21 12:43:59 [localhost] sshd[5938]: pam_krb5[5938]: TGT verified using key for 'host/mysambserver at MYDOMAIN.LOCAL'// //Jun 21 12:43:59 [localhost]

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Create the fedora-btrfs.img as an empty file. The only place this is used explicitly is tests/mountable/ test-mountable-inspect.sh, but that test already skips if !btrfs. Also this is used via guests-all-good.xml, but the script that creates this XML skips the file if it has zero size. --- test-data/phony-guests/make-fedora-img.pl | 52 +++++++++++++++-------- 1 file changed, 35 insertions(+),

Hello, I am facing 2 issues now. The first one is the more critical for me... 1. When I switch from sssd to winbind with : # authconfig --enablekrb5 --enablewinbind --enablewinbindauth --enablemkhomedir --update My sftp access did not work. Does it change the way to pass the login ? I used to connect in sftp with userlogin / userpassword //var/log/secure :// / /Jun 21 11:08:31 [localhost]

Hi all, I'm a fairly long time user of Asterisk, but I'm new to this list. I used to use the old forums some few years ago. I wanted to ask why there are different Asterisk versions, as shown by the announcements in the past week or 2: Asterisk 13.33.0 Asterisk 16.10.0 Asterisk 17.4.0 I'm currently using 16.8.0 and wondering if I should upgrade to 16.10.0, or perhaps give 17.4.0 a

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten =>

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

Hey Dave, In the case of 13 and 16, these are LTS versions which means that they get long term service. 17 is a standard release. The benefit of an LTS is that you can expect it to get bug fixes and improvements for an extended period of time without anything major being changed. If you find an LTS version that has everything you need, it's probably the safest version to choose. Any