similar to: Warnung: File dsp.c, Line 1198 ???

What does it mean ?? WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list! I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

Is any way to limit music on hold (or ringtones) to specified time ? I need it to play it ~ for 7 seconds . How to do this ? in dial plan i have: exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r when go to this extension it rings once! and then asterisk say : -- Zap/1-1 answered Modem[i4l]/ttyI0 and it stop ringing ;) becouse mean that other end is ringing :) .. BUT when the other

Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

Anybody has figured out why asterisk + snom have such bad quality using GSM? When I use GSM I see such messages dumped on asterisk console: WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect

I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hi, I am using the NUT Software 2.7.4 (Ubuntu Repository) on Ubuntu 18.04. Configuring a UPS with the snmp-ups driver and overriding the battery.runtime.low variable leads to following warnung/error: Sep 23 11:10:08 hostname snmp-ups[27115]: dstate_setflags: base variable (battery.runtime.low) is immutable Default for battery.runtime.low is 180 seconds. My idea is to increase the value to

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

hi all I'm being spammed by these messages in the console (see below) and sound doesn't work with today's cvs. I rolled back a week, and it works fine. In addition to the sound problems, I had to enable inband dtmf squelch on the dilnk mgcp phones. if not, each pressed key was counted twice NOTICE[245776]: File chan_mgcp.c, Line 710 (mgcp_rtp_read): MGCP ast_dsp_process

hi .. I have finally managed to get Chan_H323 & G729 working flawlessly, thanks to some help from Jerry McNamara. For those out there who are stuck with the same problem the procedure is : 1. install on RedHat 8.0 and nothing else (RH9 doesn't work!) 2. Install asterisk, zaptel etc. the normal way 3. Compile Pwlib & oH323 with versions taken from nufone's site

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network