search for: pcmu

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 192.16...

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 : 0x11...

...t all the successful 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...

...Agent: Grandstream SIP 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...

...dstream BT100 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 354 v=0 =1019 0 8000 IN IP4 192.168.201.111 s=SIP Call c=IN IP4 192.168.201.111 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9 a=sendrecv 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 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/8000 a=ptime:20 13 headers, 17 lines Using latest request as basis request Sending to 192.168.201.111 : 5060 (non-NAT) Found RT...

...Grandstream SIP UA 1.0.3.81 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 Fou...

...ng to 10.2.0.203 : 5060 (no NAT) Using INVITE 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-code...

...for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh " Lets change the future...lets change the world." --------------------------------- Never miss a thing. Make Yahoo your homepage. -------------- next p...

...sdp Content-Length: 246 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 Foun...

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

...6a2b3fa6f8741d4749@62.254.245.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 238 v=0 o=root 19457 19457 IN IP4 62.254.245.18 s=session c=IN IP4 62.254.245.18 t=0 0 m=audio 13270 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 62.254.245.14:5060 -- Called 3046@sip.culver-tec.com Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.254.245.18:5060;branch=z9hG4bK1fd80848 From: "2001" <sip:2001@62.254.2...

...E, REFER, SUBSCRIBE, 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 R...

...00> Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@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: "as...

...00.61.32.238 CSeq: 102 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:999528...

...v=0 o=sibtay 2890844 842807 IN IP4 192.168.0.153 s=SDP Seminar c=IN IP4 192.168.0.153 t=0 0 m=audio 13064 RTP/AVP 0 101 a=rtpmap:101 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:96 0-11,16 15 headers, 11 lines Using latest request as b...

...ng to 10.2.0.203 : 5060 (no NAT) Using INVITE 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-code...

...Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1124335166 1124335166 IN IP4 192.168.1.37 s=Polycom IP Phone c=IN IP4 192.168.1.37 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 10 lines Using latest request as basis request Sending to 192.168.1.37 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.37:2224 Found description format G729 Found d...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...the basic seting of my two GrandStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1...