similar to: Forcing Marker bit, because SSRC has changed

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

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

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

Ever since upgrading to 1.2.8 I've been getting occasional WARNINGS that say: WARNING[4356]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed. Is there something I need to fix or is this a benign message? Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.8.0/352 - Release Date: 5/30/2006

Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '0354c42214142e5d6cb8e05568c59837@10.0.2.2'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick

Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the

Hi, I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI card connected to my cisco router which is connected to FastWeb provider: does anybody knows why every time my cisco router turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio --

Greetings, upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when the PSTN party answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990

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 Javier, thanks for your reply. I already checked SIS and, while interesting, is not what I want, because: 1) it can be difficult to restore a single message/attachment from a backup 2) only the attachments, and not the entire messages, are deduped. Message-based hardlinks really exists for a reason. The good news is that I found _why_ they are not working: it depends from how dovecot and

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

Hi LLVMers, there were a gcc ICE problem discussed in current mail list. Chris was right here: http://mail.cs.uiuc.edu/pipermail/llvmdev/2003-December/000693.html saying that the PR 12544 is not really the corresponding issue :) The correct one is PR 13392: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=13392 Interesting fact is that -O2 (or -O3) goes somehow around this

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination

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

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part

Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes during voice calls after which the connection is being

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Building PowerPC.td register information header with tblgen Included from PowerPC.td:22: Parsing PowerPCInstrInfo.td:53: Variable not defined: 'GPRC'! make[3]: *** [PowerPCGenRegisterInfo.h.inc] Error 1 make[3]: Leaving directory `/pool/tmp/ssrc/llvm/lib/Target/PowerPC' maybe I just have to "make clean" and/or ./configure BTW, would it be nice to put Depend, Release and