similar to: changing of ssrc between early-media and call media

Hi Everyone, We're routing calls through Asterisk (dialing in via sip and then dialing out via SIP). We've noticed a curious behavior in chan_sip that doesn't persist with chan_pjsip. When examining the packet capture, we're seeing the SSRC changing constantly on the call. At first it happens over a variable interval (15s 6s etc) but eventually it ends up changing exactly every

The Asterisk Development Team would like to announce the release of Asterisk 13.28.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.28.0 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 16.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.5.0 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:

Dear list (and more specifically Bret), I am getting one-way (inbound only) audio when trying to place a SIP call via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out "== Forcing Marker bit, because SSRC has changed" 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and

Evnin... Googled around for this strange error meesage with no helpful results at all... Does somebody has any idea what this means? "Forcing Marker bit, because SSRC has changed" At the same time I only get inbound audio but other side can't hear me...sometimes I just hear my echo and nothing from other side... Asterisk version 1.2.9 and both participants with public IP

While sending calls to a SIP provider, the following warning generates: -- Executing Dial("SIP/1000-c317", "SIP/13057671523@209.120.202.94:5060|55|o") in new stack -- Called 13057671523@209.120.202.94:5060 -- SIP/209.120.202.94:5060-0533 is making progress passing it to SIP/1000-c317 -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317 -- Attempting

Dear list members, I am trying to use mysql for the mailbox definitions of the voicemail system. All works fine, but my asterisk does not catch the options of the mailbox. In particular, 'attach=no' has no effect. Here is my test table: mysql> select * from users where email='tit@miodominio.com';

We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10, acting as gateways from SIP to ISDN PRI interfaces. Each has one Digium TE420 (with hardware echo cancellation) and one TC400B for transcoding, in order to handle 60/90 contemporary calls in peak hours. In my logs there are hundreds of thousand warnigs per day like these: transcode.c: no samples for lintoulaw transcode.c:

I did a quick check between what I have set and your settings below. You can try the following and see if it helps In your endpoint: bind_rtp_to_media_address=yes With best regards Florian Floimair Innovation - Software-Development - VoIP & DevOps COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstra?e 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

Asterisk Project Security Advisory - AST-2019-003 Product Asterisk Summary Remote Crash Vulnerability in chan_sip channel driver Nature of Advisory Denial of Service Susceptibility Remote

The Asterisk Development Team would like to announce the release of Asterisk 17.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.0.0 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:

Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

I have a dialplan (freepbx) that plays a busy signal in-band when an extension is busy (before an Answer). Stripped down, it looks like this: exten => 1005,n,PlayTones(busy) exten => 1005,n,Busy(20) Note that there is no Answer() prior to this. Our trunk is a PRI. When I call into this extension from outside, I get about 25 seconds of ringing, followed by a hangup. Looking at the

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Hi All am running asterisk 1.8 , its a realtime asterisk.have recently noticed an issues where am getting loads of these errors SIP/blabla requested media update control 26, passing it to SIP/ProviderX this literally keeps flooding the screen untill the call is answered and then i get normal "locally bridging two channels" infact, this has caused cross-talk (3-way call) between some

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I