similar to: DAHDI error

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not know about the W ??? I think zaptel used

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri 1.4.7 and I am getting the error: -- Requested transfer capability: 0x00 - SPEECH -- Called 23/317506XXXX -- Channel 0/23, span 1 got hangup, cause 99 -- Hungup 'DAHDI/23-1' on DIALING out. Calling in seems to work just fine. Seems like everything is configured fine... system.conf

getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into

I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel

I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4, using the wct4xxp module. All operations appear normal however I noticed an error repeating occasionally on the console. [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ROSE RETURN ERROR: [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 INVOKE ID: 11 [Jan 10 13:53:05] ERROR[6906]:

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we

I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1]

Hello all, I've got an intermittent issue with my asterisk set-up, and I'm pulling my hair out! Mostly everything works fine, but I get an error once every few days that sometimes (but not always) causes asterisk to stop accepting new calls and also stop responding to commands on the console (asterisk -r). The usual errors look something like this: [Feb 19 13:09:52] ERROR[18728]

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1, Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two TE120P cards and everything was working fine. Since they needed to add a third E1 line we decided to change one of the TE120P cards with a TE210P. After the change we had a couple of crashes (server

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 and siemens hipath 4000. (2 channels to each switch) with a TE210p card setup as T1 with em_w. When the call is initiated to either switch the phone rings, when its answered then nothing... I hear no audio etc... After the timeout period the call is hung up. The phone switch 300 needs the T1 reset as the channel is not

Hi all, Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI> pri intense debug span 1 Enabled debugging on span 1 [Dec 16

Hi, I'm getting the following error from an asterisk 1.6.0.9 installation: [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler This repeats a few times, then asterisk crashes. I can't seem to locate any info on this error at all. I'm using