similar to: Asterisk stops responding to SIP/ZAP

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35

I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

Hello, my configuration is the following: a TDM400P board with an fxs and fxo daughter boards on it. I thus connect a fax to my FXS port, after having verified that this port was correctly functioning. For this, I had tried before with a simple phone, and with some basic voicemail exten scripts. Here is my simple dialplan for my fax reception: exten => 300,1,Ringing() exten =>

Hi There. I have the following setup : Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24 My problem is as follows : If I set up a very simple extensions.conf. when I dial from a fax machine, it seems as if no fax is being recognised. If I answer the call, I can hear the fax machine beeping. extensions.conf :

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

I have the following in my extensions.conf [ext-local] exten => _53XX,1,Wait(2) exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,) This is used to match inbound caller-id for my legacy PBX. It works fine for inbound calls, but not for internal SIP calls. If I call from a SIP phone that is also in [ext-local], it looks like it

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to