similar to: RE: An old problem still hanging around?

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

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)

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, 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

I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?

I'm having an oddball issue with a Polycom SoundPoint IP 500. As you can see below Asterisk thinks there are 2 SIP channels active, but show channels tells me there 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

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it rings but I get that same error when I pick up. When the skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it does this on

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

Has anyone seen this before? This channel is consistently present on both of 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

Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16 "sip show channels" Always tends to show 100+ lines such as 192.168.1.241 (None) 2e2872da-1d 00101/21507 unkn No Rx: REGISTER Never seem to go away 198 total peers on this server All devices are behind NAT Registration expirations between 30secs to 2 minutes to help keep NAT open Should I extend the

My sip show channels shows some channels active that I can not make sense out of, and they have been that way for days, so I am pretty sure they are orphans. Is there a way to show active CALLS (instead of channels) to try and determine the source? Does the output below provide any clues as to why these channels might show active? Anyone aware of related bugs? The #'s indicate original

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.

Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Thanks! Thanks! Thanks! I've got it work!!! :-) Message: 13 Date: Sun, 16 Jan 2005 12:17:21 -0000 From: "Bill Seddon" <bill.seddon@lyquidity.com> Subject: RE: [Asterisk-Users] failed to compile zaptel on redhat To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID:

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses