search for: 00102

...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 User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 172.16.116.29 2897516#15 1db77942648 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 585240b13ef 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 0244b8e668d 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#16 46d0960e602 00102/00000 g729...

...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 00102/00002 UNKN (d) 10.204.10.15 467040 5800ec-acc0 00102/00002 UNKN (d) 10.1.1.45 909617...

...out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean? Here is the output from 'sip show channels': Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 195.7.123.234 +180924402 3c3c4cee419 00102/00000 alaw No Tx: ACK 9.9.94.9 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 +0115...

Hi, I'm trying to view a list of the active calls to see if I can restart Asterisk. When I do 'sip show channels', I get a huge list like this (just a sample pasted):- 92.110.7.210 (None) 198827f2469 00102/00000 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 6b211bb04ac 00102/00000 0x0 (nothing) No Init: OPTIONS 92.108.34.153 (None) 4e05e84848b 00102/00000 0x0 (nothing) No Init: OPTIONS 92.110.7.210 (None) 7bdb88176f0 00102...

...Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0 (nothing) No Rx: REGISTER 195.189.173.10 30124189 3b37329b407 18996/00000 0x0 (nothing) No 192.168.1.13 1007 2da66c6d6a1 00102/00000 0x280100 (g729| No Tx: ACK 192.168.1.13 1005 67384261131 00102/00000 0x280100 (g729| No Tx: ACK 192.168.1.13 1010 041c9a77455 00102/00000 0x280100 (g729| No Tx: ACK 81.201.84.45 3473290576 PUM273-UMU5 00101/00102 0x100 (g729) No Rx: AC...

...can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 192.168.25.29 648 7c24869b010 00102/00000 0x2 (gsm) No Tx: ACK 192.168.25.29 648 26e8187a0a4 00102/00000 0x0 (nothing) No Tx: CANCEL 192.168.25.29 648 5289c52b77e 00102/00000 0x0 (nothing) No Tx: CANCEL 192.168.25.29 648 7a6243bc21e 00102/00000 0x0 (nothing) No Tx: CANCE...

...ralink phones have entries in 'sip show channels' that do not go away. None of the other phones do this. Is there anyway to remove these entries without restarting Asterisk? Any ideas on what could be done to prevent this? Example output: xxx.xxx.xxx.xxx 541 14dd18886d1 00103/00102 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 1e7c2fd84ab 00103/00102 0x0 (nothing) No (d) Rx: BYE xxx.xxx.xxx.xxx 546 80f99ee6-6c 00103/00104 0x0 (nothing) No Rx: BYE xxx.xxx.xxx.xxx 546 0d9b184254b 00104/00102 0x0 (nothing) No Rx: BY...

...o why these channels might show active? Anyone aware of related bugs? The #'s indicate original data changed for security reasons. vg2-inverness-co*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Msg 205.###.247.### 55566213## 318bf4d9509 00102/12193 ulaw No Rx: ACK 64.#.11.## 55570615## 16502820-b7 00101/00102 ulaw No Rx: ACK 64.#.11.## 55573378## 62fb705108b 00102/00000 ulaw No Tx: ACK 64.#.11.## 55581484## 4b7561a076b 00102/00000 unkn No (d) Tx: CANCEL 205.###.247.### 30326662## 681a2af4...

...irst, I've 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 0e00060800...

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

...ery weirds ip on the server, see here: mypbx*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 10.4.1.130 (None) b7dd744679a 00101/14365 0x0 (nothing) No Rx: REGISTER 10.4.1.151 308 8cbe459c33e 00102/16076 0x0 (nothing) No Tx: NOTIFY 10.4.1.144 368 607d5af86cd 00102/25625 0x0 (nothing) No Tx: NOTIFY 169.254.236.26 308 4f407ce65eb 00102/15097 0x0 (nothing) No Tx: NOTIFY 169.254.21.164 309 f1e31e48e10 00102/23948 0x0 (nothing) No...

...t; > > > > > > > > MATT--- > > > > > > > > > > > > > > > > > > > > > > > > > > > Example: > > > > > > > > > > > > 220.227.174.2 agent15 018a383f451 00102/00000 > > > > ulaw No Tx: > > > > > > ACK > > > > > > 220.227.174.2 agent12 049848c527c 00102/00000 > > > > ulaw No Tx: > > > > > > ACK > > > > > > 220.227.174.2 agent1 3a...

Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.173 0022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0016513973 234f7bba140 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0027226765 487b770b231 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0016513973 69b59aa2084 00102/001...

...The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e 00102/00000 ulaw 10.0.0.48 3035 0008a3d2-05 00101/00102 ulaw 10.0.0.48 3035 0feb1c11386 00103/00101 g729 65.72.107.2 5126800422 28D20837-69 00103/00101 g729 4 active SIP channel(s) 5126800422 is still on hold while 3035 talks with 8327549222. 3035 now presses...

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: 0x8CBDA...

...disallow=all allow=alaw I have also confirmed while on an outbound calls that both are using the exact same codecs. sip show channels shows pbx*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.0.55 123456789 4ea2e1314cd 00102/00000 alaw No Tx: ACK 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx: ACK 2 active SIP channels Anyone have an idea what's going on? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-use...

...IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000 00000ms 0000ms 2 150.187.xxx.yyy ildefonso C72ACD25-1A 00101/11482 00000ms 0000ms 2 2 active SIP channel(s) -- SIP/fwd-161b answered SIP/ildefonso-d2fc -- Attempting native bridge of SIP/ildefonso-d2fc and SIP/fwd-161b When it gets stablished, I get: sip show channels Pe...

...e: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 10.9.9.9 202 11a6323d2ff 00102/00000 0x100 (g729) No Tx: ACK 10.9.9.10 201 a8749-c0a80 00101/00001 0x100 (g729) No Rx: ACK Installing the open source binary of g279 (codec_g729.so in modules dir) and restarting asterisk, the same call require transcoding: Peer User/ANR Call ID...

...le and after about a minute the phone disconnects. A 'sip show channels' reveals the following: Peer User/ANR Call ID Seq (Tx/Rx) Format [VoIP-provider] [ext. number dialed] 5b1fe97c04d 00103/00000 g729 [IP of Cisco phone] [ID of Cisco] 0002b9a7-4b 00102/00102 ulaw 2 active SIP channel(s) Here g729 pops up, even though I have configured [VoIP-provider] to only allow/use ulaw/alaw. asterisk -vvv shows: -- Executing Dial("SIP/[ID of Cisco]-4663", "SIP/[VoIP-provider]/[ext. number dialed]") in new stack -- Called [Vo...

...those phones when we register with the sip-extension, we've experienced the same problem. This is the output from sip show channels: Peer User/ANR Call ID Seq (Tx/Rx) Format x.x.x.x (None) 3c267009a87 00101/79668 unknow x.x.x.x 037807490 07b9a829600 00102/00000 alaw x.x.x.x 32164695 2C47054A-25 00101/00101 alaw x.x.x.x 757 5a5525f02b1 00102/00000 ulaw x.x.x.x 755 3c278bfbb23 00101/00002 ulaw x.x.x.x 756 4a355b504b9 00102/00000 unknow(d) 6 active SIP channel(s) The extension that always re...