[Asterisk-Users] Sound issue
Hey everyone, I know this is a problem with mpg123, but it just started happening and I have no idea why. I haven't changed any of the audio format settings yet. Before tonight, I was able to call, listen to the queues, hear the music on hold, no problems. I added a new context to a dial plan, reloaded and now I get this error. Ouch ... error while writing audio data: : Broken pipe Then asterisk just crashes. I have so far tried to make clean make make install, that didn't work. Replaced my own configuration files with the samples. Still I get the same error. I also noticed that i had a lot of processes of mpg123 running. [EMAIL PROTECTED] asterisk]# pgrep mpg123 2354 2372 2390 2462 2763 2785 2805 2823 2843 2862 2883 2905 2927 2949 2966 2986 3005 8608 8656 9180 9708 I can't seem to get rid of them either. I even restarted the whole server, which asterisk is configured to auto start and they are back. Currently I am running asterisk 1.2.4. The only other thing that I changed today was add odbc support for mssql. But I don't see any correlation there. Yes, I am getting a db error, but that is because the server isn't ready to accept asterisk yet. Any suggestions? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound issue
I know this is a problem with mpg123, but it just started happening and I have no idea why. I haven't changed any of the audio format settings yet. Before tonight, I was able to call, listen to the queues, hear the music on hold, no problems. I added a new context to a dial plan, reloaded and now I get this error. Ouch ... error while writing audio data: : Broken pipe Then asterisk just crashes. I have so far tried to make clean make make install, that didn't work. Replaced my own configuration files with the samples. Still I get the same error. I also noticed that i had a lot of processes of mpg123 running. [EMAIL PROTECTED] asterisk]# pgrep mpg123 2354 2372 2390 2462 2763 2785 2805 2823 2843 2862 2883 2905 2927 2949 2966 2986 3005 8608 8656 9180 9708 I can't seem to get rid of them either. I even restarted the whole server, which asterisk is configured to auto start and they are back. Currently I am running asterisk 1.2.4. The only other thing that I changed today was add odbc support for mssql. But I don't see any correlation there. Yes, I am getting a db error, but that is because the server isn't ready to accept asterisk yet. Any suggestions? The mpg processes are a fairly common problem when asterisk has an issue. When asterisk dies, it can't clean up those processes. Just kill them with 'kill -9 2354 2372 etc' Then go back and figure out what you did for a context change that broke asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound Issue
Hey Rich and everyone. I tried what you suggested, and it didn't work. I even recomplied everything, moved all of my configuration files out and remade the samples, so as far as I can tell everything is back to day 1. However, it is still pulling in the database information. This is really the only thing I could think of that is causing any problems. Here is a list from the CLI of errors, warnings, etc. [ Booting...Feb 25 23:57:08 NOTICE[18542]: cdr.c:1188 do_reload: CDR simple logging enabled. ..Feb 25 23:57:08 ERROR[18542]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server on . Check debug for more info. Feb 25 23:57:08 WARNING[18542]: res_config_mysql.c:450 load_module: MySQL RealTime: Couldn't establish connection. Check debug. Feb 25 23:57:08 NOTICE[18542]: config.c:863 ast_config_engine_register: Registered Config Engine mysql ..Feb 25 23:57:08 NOTICE[18542]: config.c:863 ast_config_engine_register: Registered Config Engine odbc .Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:294 load_odbc_config: registered database handle 'asterisk' dsn-[asterisk] Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:552 odbc_obj_connect: Connecting asterisk Feb 25 23:57:08 WARNING[18542]: res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified Feb 25 23:57:08 NOTICE[18542]: res_odbc.c:597 load_module: res_odbc loaded. ...Warning, flexibel rate not heavily tested! .Feb 25 23:57:08 WARNING[18542]: pbx_dundi.c:4584 set_config: Unable to look up host 'voip-1.sgnwmi-1.mercury.net' Feb 25 23:57:08 WARNING[18542]: chan_mgcp.c:4213 reload_config: Unable to get our IP address, MGCP disabled ..Feb 25 23:57:08 WARNING[18542]: chan_skinny.c:3154 reload_config: Unable to get our IP address, Skinny disabled ..Feb 25 23:57:08 WARNING[18542]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_tds.so: undefined symbol: tds_free_connection Feb 25 23:57:08 WARNING[18542]: loader.c:554 load_modules: Loading module cdr_tds.so failed! Ouch ... error while writing audio data: : Broken pipe I'm not to familar with removing src files after I compile them (still a little new to unix), but is there a way I can remove the files for unixODBC and freeTDS? Otherwise, I'm lost as to what could be causing the problem. Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound issue with Asterisk
Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am in the office now so I am able to provide some more information about the issue that I am having. Here is the kernel if this helps Fedora core 4 -- 2.6.11-1.1369_FC4smp I know that ztdummy is at least loaded now. Also as stated before there is nothing plugged into the T1 card. So I wasn't sure if that was causing a problem or not which is why I enabled ztdummy but it was not the first time I e-mailed you. # lsmod | grep ztdummy ztdummy 7748 0 zaptel192516 6 ztdummy,wct4xxp If I look at the connections from tcpdump, I see my phone call coming in, but no traffic is being sent back to the phone. With an Echo() test, I see the traffic going back and forth, but when I call into a menu, then there is nothing. Thanks, Kevin I ran a sip debug as well but I felt it was better at the end of the e-mail: -- SIP read from 64.7.189.14:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK2b87d1ab;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4a36d77b To: sip:[EMAIL PROTECTED];tag=a0efbf44ecab5900 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream BT100 1.0.6.7 Contact: sip:[EMAIL PROTECTED] Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- SIP read from 64.7.189.14:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Supported: replaces Call-ID: [EMAIL PROTECTED] CSeq: 19606 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14 s=SIP Call c=IN IP4 64.7.189.14 t=0 0 m=audio 5004 RTP/AVP 2 8 4 18 15 97 9 a=sendrecv a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 --- (13 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 64.7.189.14 : 5060 (non-NAT) Found peer 'budgeTone-PubIP' Reliably Transmitting (no NAT) to 64.7.189.14:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c;received=64.7.189.14 From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED];tag=as6f00184d Call-ID: [EMAIL PROTECTED] CSeq: 19606 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=351ca5f6 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 64.7.189.14:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED];tag=as6f00184d Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 19606 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines)--- -- SIP read from 64.7.189.14:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7 From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Supported: replaces Proxy-Authorization: Digest username=budgeTone-PubIP, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=351ca5f6, response=3748b6120c7f4ecc4873cbdaf178d507 Call-ID: [EMAIL PROTECTED] CSeq: 19607 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=budgeTone-PubIP 8000 8001 IN IP4 64.7.189.14 s=SIP Call c=IN IP4 64.7.189.14 t=0 0 m=audio 5004 RTP/AVP 2 8 4 18 15 97 9 a=sendrecv a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 --- (14 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 64.7.189.14 : 5060 (non-NAT) Found peer 'budgeTone-PubIP' Found RTP audio format 2 Found RTP audio format 8 Found RTP audio
Re:[Asterisk-Users] Sound issue with Asterisk
Hi everyone and Steve, Well the problem I wrote about is fixed. Here is what I did to resolve the issue. I was running kernel 2.6.11-1.1369_FC4smp before. I went and upgraded to 2.6.14-1.1656_FC4smp along with the development files (which I finally found were not installed). After I installed the new kernel, recompiled asterisk, added my SIP and Extensions back in and all the menus in the demo, echo, etc, were all working. Thanks Steve for helping me and hopefully this will help someone out as well. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sound issue
Hi all, I'm new to asterisk. I have it installed and 1 x100p card. I'm trying to use kphone to connect out. But, when I start * it gives me sound card busy error. I've checked ps aux and nothing seems to have the sound card. Any ideas? Does starting asterisk automatically take the sound card? Also, can you use the phone connected into the XFO card w/ asterisk or do you need an XFS card or soft/hard phone? Sorry for the questions, but I can't seem to find any answers on the wiki etc. Thanks in Advance. Vlok ?@ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
