search for: unkn

Displaying 20 results from an estimated 102 matches for "unkn".

2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
...nds of devices. The Peer with IP 10.204.10.12 is an AudioCodes MP-124, and every other IP is a number of Welltech 3504A 4-port FXS devices. asterisk-srv1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 10.204.10.12 161913 2d381f58106 00103/00000 UNKN (d) 10.204.10.12 161913 1e68f3610c9 00103/00000 UNKN (d) 10.204.10.12 463913 28862156821 00102/05918 UNKN (d) 10.204.10.12 468945 25028137781 00102/16213 UNKN (d) 10.204.10.20 305129 57f9ac-acc0 00102/00002 UNKN (d) 10.204.10.15 467040 57f9cc-acc0...
2007 Feb 27
1
Help understanding SIP SHOW CHANNELS
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows "unkn" for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3. I see a lot of "WARNING[20224]: udptl.c:819 ast_udptl_new_with_bindaddr: No UDPTL ports remaining" errors - is this related to number 2 above? Thanks, MD Peer Us...
2007 Sep 06
1
Dead SIP channels
...9.94.9 6478517573 2752611-195 00101/00001 ulaw No Rx: ACK 136.59.30.19 8787041796 76775e35788 00102/00000 ulaw No Tx: ACK 9.9.95.13 9057047798 2752419-199 00101/00001 ulaw No Rx: ACK 195.7.123.234 +011503733 25afde8070b 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011503733 71688696061 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011503733 1700ab8b2ae 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011578435 0ecb33f75bb 00102/00002 unkn No (d) Rx: BYE 195.7.123.234 +011962642 71eac20715c 0...
2008 Feb 13
3
urgent-channels
Hi All I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/00000 unkn No 215.96.142.83 (None) caac0846-cf 00101/00000 unkn No 192.168.8.106(None) 94910146-46 00101/00000 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/00000 unkn No 85.219.172.253 (None) 67a0d6b3191 00101/00000 unkn No 85.219.172.253 (None) 0d778c31...
2006 Jun 22
3
Showing Current Calls
...@one_start:2> 2944093@one_start:2 Up Dial(SIP/2944093|36|tr) 2 active channels 1 active call hestia*CLI> hestia*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message xxx.yyy.128.115 (None) e77bba33-cc 00101/02261 unkn No Rx: REGISTER xxx.yyy.128.110 (None) 739f4603-e8 00101/00778 unkn No Rx: REGISTER xxx.yyy.128.86 (None) 56caad3a-eb 00101/01046 unkn No Rx: REGISTER xxx.yyy.128.115 (None) 91ea0410-60 00101/02262 unkn No Rx: REGISTER xxx.yyy.128.86...
2007 Dec 07
2
7960 Won't Register Yet Multiple Attempts?
...ebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If the phone itself is rebooted, I see a mess of registration attempts via SIP channels: 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None) 000e8XXX-5d 00101/00220 unkn No Rx: REGISTER 7X.183.246.XXX (None)...
2004 Aug 26
0
Asterisk media problem behind NAT
...application/sdp Content-Length: 148 v=0 o=par 0 0 IN IP4 <gateway1> s=- c=IN IP4 <gateway1> t=0 0 m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18 m=video 22222 RTP/AVP 26 34 31 10 headers, 7 lines Using latest request as basis request Sending to 172.16.1.54 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found video format UNKN Found video format UNKN Found video format UNKN Capabilities: us - 786446, them - 303/851968, combined - 78...
2007 Aug 17
4
Call Limits
Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards
2007 May 09
10
SIP Problems continue...
...e currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116 716 0a2a959d3d3 00102/00000 unkn No Init: INVITE 10.200.26.115 715 1dee947d485 00102/00000 unkn No Init: INVITE 10.200.26.104 704 28808764699 00102/00000 unkn No Init: INVITE 10.200.26.104 704 36d3e88f59c 00102/00000 unkn No Init: INVITE 10.200.26.104 704 0e00060800d 00102/000...
2004 Jun 03
4
miserable time with Cisco ATA186
...Answering with capability 0x10(G726) Answering with capability 0x20(ADPCM) Answering with capability 0x40(SLINR) Answering with capability 0x80(LPC10) Answering with capability 0x100(G729A) Answering with capability 0x200(SPEEX) Answering with capability 0x400(ILBC) Answering with capability 0x800(UNKN) Answering with capability 0x1000(UNKN) Answering with capability 0x2000(UNKN) Answering with capability 0x4000(UNKN) Answering with capability 0x8000(UNKN) Answering with non-codec capability 0x1(G723) 12 headers, 20 lines Reliably Transmitting: INVITE sip:8664113278@munged SIP/2.0 Via: SIP/2.0/UD...
2003 Sep 27
1
Continuing Budgetone woes
...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 audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G7...
2003 Oct 23
0
WAS: Call pickup (*8) on SIP devices. Bug #116
...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 PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 2147483647, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them...
2004 May 18
0
No luck using asterisk as proxy...
.../AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Sending to 213.208.99.115 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format tele...
2003 Sep 25
3
SIP codecs Errors
...10) iLBC 65536 (1 << 16) JPEG image 131072 (1 << 17) PNG image 262144 (1 << 18) H.261 Video 524288 (1 << 19) H.263 Video The "sip debug" show the following: *CLI> sip debug SIP Debugging Enabled Sip read: INVITE sip:2060@172.16.254.96;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060 From: "52880472" <sip:52880472@172.16.254.96> To: <sip:2060@172.16.254.96;user=phone;phone-context=unknown> Date: Thu, 25 Sep 2003 16:49:48 ARBUE Call-ID: 410A02D2-EEC811D7-8FA5ADBB-C9B38AEA@172.16.254.96 Cisco-Guid: 1091135146-4...
2004 Jul 16
1
SIP channels UNKWN
...here are no calls active. Anyone have any idea why this is happening? The Polycom occasionally stops accepting calls and requires a power cycle. fs-1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 172.16.5.126 (None) 7167456b-51 00101/00621 UNKN 172.16.5.126 (None) fd6a6881-ea 00101/00621 UNKN 2 active SIP channel(s) fs-1*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) fs-1*CLI>
2004 Sep 23
0
RE: An old problem still hanging around?
...ust run the command "sip show channels" I get a list of channels even though there is no one on the phone (we only have 4 so it's easy to tell). Here is what I get: Peer User/ANR Call ID Seq (Tx/Rx) Format 192.168.0.22 (None) 4c81ac8e90c 00101/00000 UNKN 192.168.0.22 (None) 984ee48048d 00101/00000 UNKN 192.168.0.22 (None) 200d9d37123 00101/00000 UNKN Is this normal? Why just one phone (a Grandstream Handytone ATA)? Running "sip show channel 984ee48048d" I get the output below so it seems "active":...
2007 Nov 16
1
channels to destroy
Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://...
2004 May 20
4
Mystery SIP channels
...my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN
2004 Jan 06
1
ATA call
...P 18 8 0 101 > a=rtpmap:18 G729/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 12 headers, 11 lines > Using latest request as basis request > Sending to 192.168.0.150 : 5060 (NAT) > Found audio format UNKN > Found audio format ALAW > Found audio format UNKN > Found audio format UNKN > Found description format G729 > Found description format PCMA > Found description format PCMU > Found description format telephone-event > Capabilities: us - 256, them - 268/0, combined - 256 &gt...
2003 Nov 05
0
SIP broken for budgtone.
...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 Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found descriptio...