search for: pcma

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 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.04813400...

...:16:17] DEBUG[11426] res_fax.c: channel 'SIP/SOV20001-0007cb04' using FAX session '660' [Oct 14 14:16:17] DEBUG[11426] chan_sip.c: T38 state changed to 3 on channel SIP/SDSD0005-0007cb05 PCAP text output of 1st case: 216.025063 192.168.196.3 -> 192.168.196.94 RTP PT=ITU-T G.711 PCMA, SSRC=0x6118CC28, Seq=4877, Time=1179871936 216.042133 192.168.196.94 -> 192.168.196.3 RTP PT=ITU-T G.711 PCMA, SSRC=0x205734A, Seq=8641, Time=2227760 216.045031 192.168.196.3 -> 192.168.196.94 RTP PT=ITU-T G.711 PCMA, SSRC=0x6118CC28, Seq=4878, Time=1179872096 216.062169 192.168.196.94 ->...

Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my peers shows this order correctly: Codecs...

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no): softphone 1 <- PCMA -> Asterisk <- G...

...ig and upgraded to 1.6.1 but it didnt change anything, currently running asterisk 1.4.26.1 on 64 bit intel platform with opensuse. Here is the tcpdump view from wireshark, xxx is providers ip and yyy is asterisk: 6218 207.717454 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680 6219 207.717481 yyy.yyy.yyy.yyy xxx.xxx.xxx.xxx RTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496 6220 207.737442 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54365, Time...

...P UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 257 v=0 o=btel 0 0 IN IP4 192.168.1.21 s=- c=IN IP4 192.168.1.21 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.21 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found au...

....0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 263 v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.223 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN...

...and and hope I can get some guidance on. I understand the decoding/decoupling part as it is the same as the one described in the stereo docs: From mapping0.c: /* channel coupling */ for(i=info->coupling_steps-1;i>=0;i--){ float *pcmM=vb->pcm[info->coupling_mag[i]]; float *pcmA=vb->pcm[info->coupling_ang[i]]; for(j=0;j<n/2;j++){ float mag=pcmM[j]; float ang=pcmA[j]; if(mag>0) if(ang>0){ pcmM[j]=mag; pcmA[j]=mag-ang; }else{ pcmA[j]=mag; pcmM[j]=mag+ang; } else if(ang>0){ pcmM[j]=m...

...pires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:10...

...request as basis request - 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.2.0.203:24394 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-ev...

...calls have SDP negotiation that look like this: (inside INVITE request body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=root 835643920 835643920 IN IP4 201.234.196.171 s=Asterisk PBX 11.10.0 c=IN IP4 201.234.196.171 t=0 0 m=audio 12112 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA...

...ntact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp> Supported: replaces User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (16 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.150.51:5...

...replaces Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Content-Length: 394 v=0 o=root 15363811 15363812 IN IP4 192.168.2.1 s=sipgate VoIP GW c=IN IP4 192.168.2.1 t=0 0 m=audio 7070 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --...

...46 Accept: application/sdp Remote-Party-ID: "5285" <sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84 s=SIP Call c=IN IP4 192.168.1.84 t=0 0 m=audio 31790 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 192.168.1.84 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format...

...CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 16 Apr 2004 19:21:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 258 v=0 o=root 14316 14316 IN IP4 10.1.0.11 =sessionI> c=IN IP4 10.1.0.11 t=0 0 m=audio 18406 RTP/AVP 8 0 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.119:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357 From: "asterisk" <sip:asterisk@10.1.0.11>;ta...

...NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Fo...

...9bc1d8dd93a400263c775c63f5b@192.168.1.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 236 v=0 o=root 3296 3296 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 10860 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 213.137.73.140:5060 -- Called 16507148980@iconnect Retransmitting #1 (no NAT): INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK41236072 From: "asterisk" <sip:...

...2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.238 s=session c=IN IP4 200.61.32.238 t=0 0 m=audio 15740 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP Oct 22 09:19:49.938: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.61.32.238:5060;branch=z9hG4bK67299947 From: "52880472" <sip:52880472@200.61.32.238>;tag=as7493951b To: <sip:99952880474@200.61.32.142&g...

...request as basis request - 001d45b6-1d490087-44d5cce8-d599695a at 10.2.0.203 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.2.0.203:29422 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-ev...