[asterisk-users] Same Problem with AEX808E Re: No incoming audio on Dahdi channels (TDM410P)
device 8005 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at cc00 Memory at dfdffc00 (32-bit, non-prefetchable) Expansion ROM at dfdc [disabled] Capabilities: [c0] Power Management version 2 Is that normal? Here's the output of dahdi_diag 1: dahdi: Dump of DAHDI Channel 1 (WCTDM/0/0,1,1): dahdi: flags: 201 hex, writechunk: ee0d008c, readchunk: ee0d0098 dahdi: rxgain: f8b8c480, txgain: f8b8c480, gainalloc: 0 dahdi: span: e9460054, sig: 2004 hex, sigcap: 6085 hex dahdi: inreadbuf: -1, outreadbuf: -1, inwritebuf: -1, outwritebuf: -1 dahdi: blocksize: 0, numbufs: 2, txbufpolicy: 0, txbufpolicy: 0 dahdi: txdisable: 0, rxdisable: 0, iomask: 0 dahdi: curzone: , tonezone: 0, curtone: , tonep: 0 dahdi: digitmode: 0, txdialbuf: , dialing: 0, aftdialtimer: 0, cadpos. 0 dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0 dahdi: ec: , echocancel: 0, deflaw: 0, xlaw: f8b6f2a0 dahdi: echostate: 00, echotimer: 0, echolastupdate: 0 dahdi: itimer: 0, otimer: 0, ringdebtimer: 0 No idea what any of that means or how it's relevant. dmesg is full of interrupt misses and polarity reversals: ... wctdm24xxp0: Missed interrupt. Increasing latency to 18 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 19 ms in order to compensate. 29794979 Polarity reversed (1 - -1) 29795839 Polarity reversed (-1 - 1) wctdm24xxp0: Missed interrupt. Increasing latency to 20 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 21 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 22 ms in order to compensate. 31595924 Polarity reversed (1 - -1) 31596867 Polarity reversed (-1 - 1) ... RING on 1/2! 74920374 Polarity reversed (-1 - 1) NO RING on 1/2! 74921961 Polarity reversed (1 - -1) RING on 1/2! NO RING on 1/2! NO BATTERY on 1/2! BATTERY on 1/2 (-)! Running AsteriskNow 1.5. X Windows is disabled. Ideas? Suggestions? Thoughts? Going to build another PC and toss this in there to see what happens tonight. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:2...@tiendalinux.com Email/MSN: nes...@tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BUG in Asterisk 1.6.1.0 and issue in DAHDI 2.1.0.4
Starting playing with asterisk 1.6.1.0 i found the following problems:
In the cdr_pgsql, the sql statement is wrong:
2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr
(accountcode,calldate,src,clid,dst,src,dcontext,dst,clid,dcontext,channel,channel,dstchannel,lastapp,lastapp,lastdata,lastdata,duration,start,billsec,answer,disposition,end,duration,amaflags,billsec,accountcode,disposition,uniqueid,amaflags,userfield,userfield,uniqueid)
VALUES ('4868','2009-06-25 12:14:11','unknown','bga-gw-1-4
unknown','4','unknown','from-bga-gw','4','bga-gw-1-4
unknown','from-bga-gw','SIP/bga-gw-1-4-09be9960','SIP/bga-gw-1-4-09be9960','DAHDI/5-1','Dial','Dial','DAHDI/g11/4868','DAHDI/g11/4868',170,'2009-06-25
12:14:11',170,'2009-06-25 12:14:11','ANSWERED','2009-06-25
12:17:01',170,3,170,'4868',8,'1245950051.272',3,'4','4','1245950051.272')
2009-06-25 12:17:01 COT ERROR: column src specified more than once
and asterisk shows:
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:309 pgsql_log: Failed to
insert call detail record into database!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:310 pgsql_log: Reason:
ERROR: column src specified more than once
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:311 pgsql_log: Connection
may have been lost... attempting to reconnect.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:314 pgsql_log: Connection
reestablished.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:320 pgsql_log: HARD ERROR!
Attempted reconnection failed. DROPPING CALL RECORD!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:321 pgsql_log: Reason:
ERROR: column src specified more than once
Also in chan_dahdi i got the following:
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: k
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: o
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: w
Slds.
--
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:2...@tiendalinux.com
Email/MSN: nes...@tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia
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[asterisk-users] Documentation of users.conf
Hello, does anybody know where is documented every parameter of the users.conf file in the asterisk distribucion tarball ? thanks a lot. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with Asterisk to Huawei SoftX3000 registry problem
Hello Asterisk People, I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i can succesfully connect other softphones like Zoiper, but when it comes to Asterisk SIP Client, the system doesn't authenticate, i have the following configuration: peer: 10.220.0.2 username: 4857768 password: 4857768 the configuration is as follows: in the general section: register = 4857768:[EMAIL PROTECTED] in the sip client / peers section: [epmbog] type=peer host=10.220.0.2 secret=4857768 username=4857768 canreinvite=no [Aug 19 06:48:28] NOTICE[7608]: chan_sip.c:12525 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) i have tested numerous variants, but the huawei refuse to authenticate the asterisk. i attache the logs producted with Zoiper, and the one produced with Asterisk. Any help will be greatly apreciated. Thanks. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia REGISTER sip:10.220.0.2;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK-d8754z-2bc2d270485ea817-1---d8754z-;rport Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;rinstance=40f3e87db666ccb3;transport=UDP To: sip:[EMAIL PROTECTED];transport=UDP From: sip:[EMAIL PROTECTED];transport=UDP;tag=9d083153 Call-ID: OWNlMzM5MzcyY2RjZWE3ODI5YmNjYzg1OGUyOGM0OWM. CSeq: 1 REGISTER Expires: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO User-Agent: Zoiper for Windows rev.650 Allow-Events: presence Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK-d8754z-2bc2d270485ea817-1---d8754z-;rport=5060 To: sip:[EMAIL PROTECTED];tag=4d78afba;transport=UDP From: sip:[EMAIL PROTECTED];tag=9d083153;transport=UDP Call-ID: OWNlMzM5MzcyY2RjZWE3ODI5YmNjYzg1OGUyOGM0OWM. CSeq: 1 REGISTER Server: Huawei SoftX3000 V300R006 WWW-Authenticate: Digest realm=huawei, nonce=2c5e846455ff57f351423612f73406c3,domain=sip:huawei.com, stale=false,algorithm=MD5 Content-Length: 0 REGISTER sip:10.220.0.2;transport=UDP SIP/2.0 Via: SIP/2.0/UDP ;branch=z9hG4bK-d8754z-30acd274a9144d25-1---d8754z-;rport Max-Forwards: 70 Contact: sip:4857768;rinstance=40f3e87db666ccb3;transport=UDP To: sip:[EMAIL PROTECTED];transport=UDP From: sip:[EMAIL PROTECTED];transport=UDP;tag=9d083153 Call-ID: OWNlMzM5MzcyY2RjZWE3ODI5YmNjYzg1OGUyOGM0OWM. CSeq: 2 REGISTER Expires: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO User-Agent: Zoiper for Windows rev.650 Authorization: Digest username=4857768,realm=huawei,nonce=2c5e846455ff57f351423612f73406c3,uri=sip:10.220.0.2;transport=UDP,response=a1b246ef1363bf8592fb5114dcc32d1d,algorithm=MD5 Allow-Events: presence Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK-d8754z-30acd274a9144d25-1---d8754z-;rport=5060 Contact: sip:[EMAIL PROTECTED];user=phone;expires=70 To: sip:[EMAIL PROTECTED];tag=d7d9f98c;transport=UDP From: sip:[EMAIL PROTECTED];tag=9d083153;transport=UDP Call-ID: OWNlMzM5MzcyY2RjZWE3ODI5YmNjYzg1OGUyOGM0OWM. CSeq: 2 REGISTER Expires: 70 Server: Huawei SoftX3000 V300R006 Content-Length: 0 REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.220.0.2:5060: REGISTER sip:epmbog SIP/2.0 Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK637e5232;rport From: sip:[EMAIL PROTECTED];tag=as26053559 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- chamber*CLI --- SIP read from 10.220.0.2:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK637e5232;rport=5060 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=as26053559 To: sip:[EMAIL PROTECTED];tag=3abcb6da CSeq: 102 REGISTER WWW-Authenticate: Digest realm=huawei, nonce=de2b14c186143b68324075e0939be2b7,domain=sip:huawei.com, stale=false,algorithm=MD5 Server: Huawei SoftX3000 V300R006 Content-Length: 0 - --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name epmbog REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 10.220.0.2:5060: REGISTER sip:epmbog SIP/2.0 Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK33a68168;rport From: sip:[EMAIL PROTECTED];tag=as73522e9a To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=4857768, realm=huawei, algorithm=MD5, uri=sip:epmbog, nonce=, response=9617be9b31db41c1742f43d72a9fba11 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- chamber*CLI --- SIP read from 10.220.0.2:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.222.4.130:5060;branch=z9hG4bK33a68168;rport=5060 Call-ID: [EMAIL PROTECTED] From: sip
Re: [asterisk-users] Help with Asterisk to Huawei SoftX3000 registry problem
Philipp Kempgen wrote: Did you enable pedantic=yes in sip.conf? thank you very much for your help, it fix the problem. Is there any other issue that i have to take in mind for placing calls ? is there any option for set up pedantic for selected peers ? i use broadvoice too and it requires pedantic=no on the configuration: http://www.broadvoice.com/support_install_asterisk.html , how can coexist this two peers on just one asterisk ? [general] context=from-sip; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls pedantic=yes; Enable checking of tags in headers, language=es ; Default language setting for all users/peers dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity limitonpeers = yes ; Apply call limits on peers only. This will improve #include custom.d/sip_general.conf [authentication] #include custom.d/sip.conf -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Asterisk to Huawei SoftX3000 registry problem
I have tested pedantic=yes just in a peer configuration and it don't work, is a global setting. (tested with asterisk 1.2.21) slds. Philipp Kempgen wrote: Alex Balashov schrieb: [pedantic] is peer-specific in this context, so it's just like any other option that is also particular to certain peers. Really? I don't think so. Grüße, Philipp Kempgen -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] StatusComplete is getting me sick !!
Hello Asterisk People. Asterisk have a really annoying bug, i use frequently the manager status command and when asterisk decide not to show the statuscomplete event, it really don't show the statuscomplete string, in fact none of the AgentsComplete, QueuesComplete' are shown I use it for monitoring a queue, but this is really getting me sick. Does anybody have to deal with this issue and found a solution ? something that doesn't rely on restarting the asterisk server, since this is not a viable mechanism slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] StatusComplete is getting me sick !!
Anthony Francis wrote: Did you recently upgrade? If so, from what version to what version? I use 1.4.18.1 under debian. Thanks. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Vieri wrote: Did you try a show channels to see if there were stale channels for peer 200? I had the same problem you describe but it was due to hung channels (used * 1.4.18.1 with rtp*timeout and saw inuse peers during the pre-timeout periods even though the agents weren't on a call). No, i don't , but how do do you fix this problem ? with rtp timeout ? Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp traffic is not passing thought asterisk, or i have to put canreinvite=no ? slds. rtp*timeout for sip peers is not a fix but a workaround. Try to set both values and reload sip. Then when you witness what you posted try doing a core show channels. You can then try to soft hangup a stuck channel or wait for the rtp*timeouts. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Hello Asterisk People, I have two annoying bugs in asterisk, that i want to know if some of you have already found a way to fix: Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch. 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Not so nice thing to do 2. AgentCallBack I know i shouldn't have to use this function, since it is deprecated but lets comment the behavior Everything works fine, but when there are calls waiting in the queue, and the agent log in using this function, the agent is able to take the call , but the system log off immediately after the agent hang up the call. No solution at the moment, just login in and log in until there are no waiting calls, for the agent to not be kicked off. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface Status Bug and Re: Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
I forget another bug, i use the asterisk manager interface. I frequently use the status function but it doesn't work as expected, i use a program to parse the output of the status command but it don't behave as expected, because i always wait for the latest package: StatusComplete, and this package never arrives in the stream sometimes StatusComplete is shown, sometimes not, but when it decide not to show StatusComplete, asterisk really don't show StatusComplete, so sad for me... Temporarily workaround: put a timeout on the socket read function to assume the asterisk manager is not working properly. Well if anybody have found some solution to this i will appreciate your comments. Slds. Nestor A. Diaz wrote: Hello Asterisk People, I have two annoying bugs in asterisk, that i want to know if some of you have already found a way to fix: Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch. 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Not so nice thing to do 2. AgentCallBack I know i shouldn't have to use this function, since it is deprecated but lets comment the behavior Everything works fine, but when there are calls waiting in the queue, and the agent log in using this function, the agent is able to take the call , but the system log off immediately after the agent hang up the call. No solution at the moment, just login in and log in until there are no waiting calls, for the agent to not be kicked off. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Mojo with Horan Company, LLC wrote: Nestor A. Diaz wrote: 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Does a simple sip reload work, or do you really need to go to all the trouble of removing the peer definition? sip reload doesn't work, that's what i have to remove the peer definition, reload, recreate and reload. slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device::
Walter Willis wrote: not work fine Actually it is recognized as an x100p device: Nov 21 19:54:34 asterix kernel: Zapata Telephony Interface Registered on major 196 Nov 21 19:54:34 asterix kernel: PCI: Found IRQ 5 for device 00:0d.0 Nov 21 19:54:34 asterix kernel: wcfxo: DAA mode is 'FCC' Nov 21 19:54:34 asterix kernel: Found a Wildcard FXO: Wildcard X100P Nov 21 19:54:34 asterix kernel: Registered tone zone 0 (United States / North America) i have been able to call from the outside and the default greeting sound good, but it can not recognize tones, when program the extension to dial an inside line the sound is very bad, too much noise !!! i think the problem is with full duplex. it will be nice to investigate if we can modify the sources to make this chipset (62802-52) work with asterisk in a nice way, i have been dealing with rxgain and txgain in order to tune the card, but i have failed, the sound is still bad. 62802 is one of the chipset that it is still available on the market, it is not designed to compete against digium analog card, is designed to introduce people on the voip field, for this it is important to be supported, think of PC vs. Apple, the more people will use Asterisk the best the business will become. Somebody have deal with zapata sources in order to make some changes and make that chipset works ? does anyone have tried newer intel modem chipset with asterisk ? they work ? the only chipset that works for me was the ambient md3200, have some echo problems but with echo chancelation and training things get better after a few seconds. What are the requeriments for a modem chipset to be supported on asterisk ? p.d. i am searching for ambiend md 3200 cards, anybody know where i can buy them ? at a reasonable price off course. Thanks everyone. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NMI issues...
Hello everybody, i have found a message on the list regarding this problem, i experienced this too, my machine show this on the kernel log: Jan 15 11:21:08 catwoman kernel: Uhhuh. NMI received for unknown reason 31. Jan 15 11:21:08 catwoman kernel: Dazed and confused, but trying to continue Jan 15 11:21:08 catwoman kernel: Do you have a strange power saving mode enabled? Jan 15 11:21:08 catwoman kernel: Uhhuh. NMI received for unknown reason 21. Jan 15 11:21:08 catwoman kernel: Dazed and confused, but trying to continue Jan 15 11:21:08 catwoman kernel: Do you have a strange power saving mode enabled? and loops forever, i see some posts on the list and the solution was to start linux with the parameter nmi_watchdog=1: so put in lilo.conf: append=nmi_watchdog=1 If i use the nmi_watchdog=0 the behavior was the normal, (a lot of output on the console), with nmi_watchdog=1 the machine freeze, and with nmi_watchdog=1 the asterisk is working, however there are still a lot of nmi interrupts, but no log and it works anyway: catwoman:~# cat /proc/interrupts CPU0 0:7938515IO-APIC-edge timer 1: 1439IO-APIC-edge keyboard 2: 0 XT-PIC cascade 4: 77IO-APIC-edge serial 8: 1IO-APIC-edge rtc 12: 19IO-APIC-edge PS/2 Mouse 14: 2IO-APIC-edge libata 15: 559174IO-APIC-edge libata 18:1978930 IO-APIC-level eth0 21: 79324605 IO-APIC-level wcfxo 22: 79325021 IO-APIC-level wcfxo 26: 238520 IO-APIC-level eth1 NMI: 20571235 LOC:7938676 ERR: 0 MIS: 0 The machine is a Dell Poweredge 700 with a sata disk. Slds. -- Nestor A. Diaz Ingeniero de Sistemas y Comp. Tel. +57 1 6005490 x 211 Cel. +57 315 8190760 [EMAIL PROTECTED] http://www.tiendalinux.com Bogota, Colombia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100p to X-lite works but x-lite to x-lite not (can not transmit audio)
Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some instructions on the net, and i still have not found the answer, in conclusion: I have two x-lite clients, that can call each other, connection is stablished but no audio is transmited, i follow the recomendations: 1. Install the iblc and spx registry patch (Windows 2K) 2. Work only with the alaw codec 3. Disable silence suppresion. but i still get: RFC3389 support incomplete. Turn off on client if possible RFC3389: 5 bytes, level 0... RFC3389: 5 bytes, level 0... The above message also is showing when the call is comming from a zap defice and the application Dial (Zap, SIP/313) is executed (without the RFC3389: 5 bytes, level 0...) but it works this way. I run asterisk from the command line as user asterisk like this: asterisk -vgcd This is my sip.conf: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [312] type=friend username=312 secret=123456 host=dynamic disallow=all allow=alaw context=from-sip [313] type=friend username=313 secret=123456 host=dynamic disallow=all allow=alaw context=from-sip The extensions.conf: [from-sip] exten = 312,1,Dial(SIP/312,10) exten = 312,2,Voicemail(u312) exten = 312,102,Voicemail(b312) exten = 312,103,Hangup exten = 313,1,Dial(SIP/313,10) exten = 313,2,Voicemail(u313) exten = 313,102,Voicemail(b313) exten = 313,103,Hangup Voicemail works, but i can not leave a message from a sip phone: an 7 08:25:32 WARNING[393234]: app.c:615 ast_play_and_record: No audio available on SIP/313-47b0?? -- User hung up Urgent handler but i can do that from a zap device. I use asterisk debian's packages from testing. ii asterisk 1.0.2-2Open Source Private Branch Exchange (PBX) ii asterisk-doc 1.0.2-2Documentation for asterisk ii asterisk-sound 1.0.2-2Sound files for asterisk I like to have the x-lite clients working, any help will be apreciated. Thanks you very much for your time. -- Nestor A. Diaz LizarazoTel. +57.1.6005490 Ingeniero de Sistemas y Comp.Cel. 315 8190760 [EMAIL PROTECTED] http://soporte.tiendalinux.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
