[asterisk-users] Re: Asterisk + Huawei
Hi, that was the asterisk debug and the Huawei returns the same errors. The Huawei debug doesnt say too much, just the answer with the same errors. As you can see there the INVITE - ACK negotiation between asterisk and Huawei is there until the call is established. The errors must be in the asterisk side somewhere. Additionally this is my sip.conf.[general]bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; Address to bind tortptimeout=120disallow=allallow=ulawallow=alawmaxexpirey=500defaultexpirey=500rtptimeout=120;insecure=verycanreinvite=nocontext=default ; Default context for incoming callslanguage=en ; Default language setting for all users/peersdtmfmode=rfc2833;relaxdtmf=yes ; Relax dtmf handlingnat=no ; Global NAT settings (Affects all peers and users)register=4875129:[EMAIL PROTECTED]/1234[epmbogota]type=usercontext=default;username=4875129;secret=cvcoldeny=0.0.0.0/0.0.0.0permit=10.220.0.2/255.255.255.255host=10.220.0.2nat=no[epmbogota]type=peer;register=yesusername=4875129secret=cvcolinsecure=invitehost=10.220.0.2;fromdomain=huawei.comfromuser=4875129context=default;nat=yes;trustrpid=yes[1234]type=friendusername=1234;secret=1234;insecure=invitehost=dynamiccontext=outgoingcallerid=4875129;callerid=6055001qualify=nodtmfmode=rfc2833mailbox=4875129nat=noIm using G711 codec on both sides.Thanks a lot for your help.Ma Zhiyong [EMAIL PROTECTED] wrote: need debug * and Huawei, not * and client Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --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] Re: Asterisk + Huawei
Thanks for your answer, here is some more debug information, if is a codec interrupt issue, how can i fix it?My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The Asterisk IP address is 10.223.6.98.The Sipura is registered to the Asterisk box and the Asterisk box is registered to the Huawei softswitch. Thanks a lot for your help,Carlos Andres Medina--- INCOMING -- -- Executing Macro("SIP/10.220.0.2-08191e48", "incoming|SIP/1234") in new stack -- Executing Dial("SIP/10.220.0.2-08191e48", "SIP/1234|30") in new stackWe're at 10.223.6.98 port 19404Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDP13 headers, 11 linesReliably Transmitting (no NAT) to 10.223.6.99:5150:INVITE
sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Thu, 19 Oct 2006 01:56:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 236v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 19404 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -- Called 1234-- SIP read from 10.223.6.99:5150:SIP/2.0 100 TryingTo: sip:[EMAIL PROTECTED]:5150From: "Anonymous"
sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)- SIP read from 10.223.6.99:5150:SIP/2.0 180 RingingTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Server: Sipura/SPA2000-2.0.10(e)Content-Length: 0--- (8 headers 0 lines)--- -- SIP/1234-08197388 is ringingTransmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 180 RingingVia: SIP/2.0/UDP
10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From: Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Length: 0-- SIP read from 10.223.6.99:5150:SIP/2.0 200 OKTo: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0From: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0Call-ID: [EMAIL PROTECTED]CSeq: 102 INVITEVia: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872Contact: sip:[EMAIL PROTECTED]:5150Server: Sipura/SPA2000-2.0.10(e)Content-Length: 229Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFERSupported:
x-sipuraContent-Type: application/sdpv=0o=- 78549 78549 IN IP4 10.223.6.99s=-c=IN IP4 10.223.6.99t=0 0m=audio 21101 RTP/AVP 8 100 101a=rtpmap:8 PCMA/8000a=rtpmap:100 NSE/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=sendrecv--- (12 headers 12 lines)---Found RTP audio format 8Found RTP audio format 100Found RTP audio format 101Peer audio RTP is at port 10.223.6.99:21101Found description format PCMAFound description format NSEFound description format telephone-eventCapabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)list_route: hop: sip:[EMAIL PROTECTED]:5150set_destination: Parsing sip:[EMAIL PROTECTED]:5150 for address/port to send toset_destination: set destination to
10.223.6.99, port 5150Transmitting (no NAT) to 10.223.6.99:5150:ACK sip:[EMAIL PROTECTED]:5150 SIP/2.0Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rportFrom: "Anonymous" sip:[EMAIL PROTECTED];tag=as448023d0To: sip:[EMAIL PROTECTED]:5150;tag=e2a724add55f408bi0Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0--- -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48We're at 10.223.6.98 port 15322Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 10.220.0.2:5061:SIP/2.0 200 OKVia: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2From:
Anonymoussip:[EMAIL PROTECTED];tag=961d1a68To: sip:[EMAIL PROTECTED];user=phone;tag=as40afbad8Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Type: application/sdpContent-Length: 233v=0o=root 1760 1760 IN IP4 10.223.6.98s=sessionc=IN IP4 10.223.6.98t=0 0m=audio 15322 RTP/AVP 0 8 97a=rtpmap:0
[asterisk-users] Asterisk + Huawei
Hi everyone,Im having some troubles getting work Asterisk as SIP Client and a Huawei softswitch as SIP server. I already got my asterisk registered to the Huawei. Im working with a Sipura SPA 2000 registered to Asterisk. When im trying to make an incoming call from the Huawei to asterisk it rings but when i answered the call drp down inmediatly. The sip debug finally show this message:Reason: Q.850;cause=100;text="Invalid information element contents"And when im trying to make an outgoing call i get the following: SIP/2.0 503 Service Unavailable Reason: Q.850;cause=100;text="Message not compatible with call state or message type non-existent or not implemented"Everything is on the same network so i dont have any nat issues.Thanks a lot for your help.Carlos Andres Medina Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --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] Testing with X101P
Hi there, im testing my asterisk box using a Modem Intel 56K which on the documentation says it must have the same behavior as an X101P. So im trying to configure just a simple line with 6 extensions. Asterisk loads fine and when im testing an incoming call the welcome message answers but when im trying to dial to any extension, anything happens is like asterisk dont recognice any digit after the welcome. Im using Asterisk version 1.0.9 and im attaching my extensions.conf which is very simple. Any clue will be very helpful. Thanks a lot for your help. Carlos Andres Medina __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com extensions.conf Description: 3949034846-extensions.conf ___ --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
[Asterisk-Users] RxFAX problem
Hi there, i installed RxFAX/TxFAX with some troubles but i did it, so i have some problems when i try to receibe a FAX, i got this error:
" Executing RxFAX("IAX2/[EMAIL PROTECTED]/16385", /var/spool/asterisk/incoming/16227743.tif") in new stack
Dec 27 15:30:54 NOTICE[1141895616]: channel.c:1731 ast_set_read_format: Unable to find a path from ALAW to UNKN
Dec 27 15:30:54 WARNING[1141895616]: app_rxfax.c:264 rxfax_exec: Unable to restore read format on 'IAX2/[EMAIL PROTECTED]/16385' "
This server is using IAX with other PBX Box in other place, both of them are using G729 for IAX conection.
Any help would be useful.
Thanks for your help
Carlos Andres Medina
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[Asterisk-Users] RxFAX compile problem
Hi there, I have installed RxFAX/TxFAX with Asterisk CVS 02/16/04 and it works just fine, i installed it successfully. Now i have Asterisk CVS 10/08/04, i installed spandsp-0.0.1 with no errors. When i reinstalled asterisk i got the following error: In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init': ../include/asterisk/lock.h:311: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:311: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:311: for each function it appears in.) app_rxfax.c: In function `phase_e_handler': app_rxfax.c:86: structure has no member named `callerid' make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 I dont know what im missing, i have libtiff 3.5.7-11 Thanks for your help." Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less.___ 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] Codec Negotiation Problem
Hi there, i had installed on all my servers the codec_g729b which is the old voiceage, so a month ago i updated the codec to codec_g729a. After that i started to get this message on my asterisk console: Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Im only using g729 on all my servers, i have some IAX conections with only g729 too. I think its a negotiation problem but with the old version i didnt have this error. If anyone knows how to fix this would be very helpful. Thanks for your help Carlos Andres Medina Do you Yahoo!? Send holiday email and support a worthy cause. Do good.___ 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
[Asterisk-Users] Interrupts failure on T100P
Hi there, i have an asteriskbox working with a T100P card and its working fine. The problem issometimes i cant receiveincoming calls, all the calls are busy. When i check the machine's Hardware interrupts with the command "cat /proc/interrupts" i see no changes in the t1xxp card interrupts. So to fix the problem i have to restart the hole machine. Anybody knows why the card suddenly stop to send interruptions? the T100P has only 2 months on production and is the second time that i change it. Thanks for your help Carlos Andres Medina.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ 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
RE: [Asterisk-Users] Interrupts failure on T100P
no the card its not sharing tha interrupts with anything elseSteve Frank [EMAIL PROTECTED] wrote: Carlos Wrote: To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Interrupts failure on T100P Anybody knows why the card suddenly stop to send interruptions?the T100P has only 2 months on production and is the second time that ichange it.Is the card sharing an interrupt with anything else?___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Meet the all-new My Yahoo! Try it today! ___ 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
[Asterisk-Users] Segmentation Fault TDM22B TDM04B
Hi all, i have installed two digium cards on my asterisk box a TDM04B TDM22B. The channels are configured as show below: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)Channel 05: FXS Kewlstart (Default) (Slaves: 05)Channel 06: FXS Kewlstart (Default) (Slaves: 06)Channel 07: FXS Kewlstart (Default) (Slaves: 07)Channel 08: FXS Kewlstart (Default) (Slaves: 08) 8 channels configured. When i load the cards everything its fine, and the status of both cards is OK. The leds are green except the two FXO ports. The problem is when i try to load asterisk appears a segmentation fault, here is the error: "Sep 21 10:40:47 WARNING[1074404032]: chan_zap.c:7658 setup_zap: Ignoring faxdetectSep 21 10:40:47 WARNING[1074404032]: chan_zap.c:665 zt_open: Unable to specify channel 1: No such deviceSep 21 10:40:47 ERROR[1074404032]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such devicehere = 0, tmp-channel = 1, channel = 1Sep 21 10:40:47 ERROR[1074404032]: chan_zap.c:7377 setup_zap: Unable to register channel '1-2'Sep 21 10:40:47 WARNING[1074404032]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap'Segmentation fault" Thanks for your help. Carlos Andres Medina Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.___ 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
[Asterisk-Users] wcfxo load problem
Hi all, i have an Asterisk server with the following hardware configuration: - 2 ethernet cards (e100, 8139too) - 1 TDM22B : 2 Ports FXS 2 Ports FXO - 1 TDM04B: 4 Ports FXO bundle When i load the FXS module (modprobe wcfxs) its just fine, but when i load the FXO module (modprobe wcfxo) appears the following error: "/lib/modules/2.4.20-31.9/misc/wcfxo.o: init_module: No such deviceHint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg/lib/modules/2.4.20-31.9/misc/wcfxo.o: insmod /lib/modules/2.4.20-31.9/misc/wcfxo.o failed/lib/modules/2.4.20-31.9/misc/wcfxo.o: insmod wcfxo failed" Maybe its a hardware conflict with the IRQ, i dont know if there is a way to change the IRQ parameters and if so how can i do it? Thanks for your help. Carlos Andres Medina Do you Yahoo!?vote.yahoo.com - Register online to vote today!___ 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
[Asterisk-Users] Configure an TDM04B TDM22B
Hi, i have 2 digium cards: - TDM04B: 4 Ports FXO - TDM22B: 2 Ports FXS 2 Ports FXO when i load the module (modprobe wcfxs) i get the following error: "ZT_CHANCONFIG failed on channel 7: Invalid argument (22)Did you forget that FXS interfaces are configured with FXO signallingand that FXO interfaces use FXS signalling?" My zaptel.conf is: fxsks=1-6fxoks=7-8 # Standard for all the configurations loadzone = usdefaultzone = us My zapata.conf is [channels] ; CONFIGURACION PARA TARJETA TDM04Bsignalling=fxs_kscontext=incomingechocancel = yesechocancelwhenbridged=yesbusydetect=yesmusiconhold=defaultgroup=1channel=1-6 ; TDM04B/1callprogress=no signalling=fxo_kscontext=incomingechocancel = yesechocancelwhenbridged=yesbusydetect=yesmusiconhold=defaultgroup=2channel=7-8 ; TDM04B/1callprogress=no I dont know if there is something wrong, thanks for your help Carlos Andres Medina Do you Yahoo!?vote.yahoo.com - Register online to vote today!___ 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
[Asterisk-Users] Using Soxmix on extensions.conf
Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this:
exten = 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)exten = 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)exten = 407,3,Dial(SIP/407|20|t)exten = 407,4,System(soxmix ${MONITORDIR}/${TIMESTAMP}.${CALLERIDNUM}-in.wav ${MONITORDIR}/${TIMESTAMP}.${CALLERIDNUM}-out.wav ${MONITORDIR}/${CALLERIDNUM}.wav)exten = 407,5,Hangup
It creates the 2 files but dont do the mix between them. I dont know what the problem is.
Thanks for your help.
Carlos Andres Medina__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
[Asterisk-Users] Codec G729 Registration problem
Hi, i have a problem trying to register the codec G729, my licence is valid but when i try to Register i got the following error: "Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient which contains your machine signature and that you must send toVoiceage to obtain a valid certificate for the g729 library." My internet conection is fine and i dont have a proxy. I sent the file to voiceage support but i dont have received any answer. Thanks for your help. Carlos Andres Medina Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers!
[Asterisk-Users] Unify Incoming and Outgoing sound files
Hi, i have a call center which receives many calls at day. Those calls are stored in a directory in my asterisk server as WAV files. The problem is that each call is divided in 2 files: an IN.WAV file and OUT.WAV file. TheOUT.WAV file is what im speaking to other person, the IN.WAV file is what that person is speaking to me. I need to unify that files into one complete file that consolidate acomplete call, i dont know if there is an application that can do that. Thanks for your help Carlos Andres Medina. Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!
[Asterisk-Users] How can i get the last codec_g729.so
Hi there, im having some problems with my asterisk box, it seems codec is the principal cause of it. Reading in some forums i found that i can get the new codec_g729 from ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i checked it but the directory new_codec_binary doesnt exist. Anybody knows where can i found it?? Thanks for your help. Carlos Andres Medina Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!
[Asterisk-Users] Asterisk voicemail problem
Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box,it enters inmediatly to voicemail and then hungs up. After that its necessary tostopthe service and putting up again manually.
Here is a piece of my log file when a call is trying to incoming:
"Jun 9 06:30:16 NOTICE[1125329728]: chan_sip.c:4879 handle_response: Peer '1366' is now REACHABLE!Jun 9 06:30:31 WARNING[1125329728]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M -- Executing Goto("Zap/1-1", "4222760|s|1") in new stack^M -- Goto (4222760,s,1)^M -- Executing BackGround("Zap/1-1", "welcome-4222760") in new stack^M -- Accepting call from '16227735' to '4222760' on channel 1, span 1^M -- Playing 'welcome-4222760' (language 'en')^M -- Executing BackGround("Zap/1-1", "menu-4222760") in new stack^M -- Playing 'menu-4222760' (language 'en')Jun 9 06:30:42 WARNING[1125329728]: chan_sip.c:49
5
retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M == CDR updated on Zap/1-1^M -- Executing Dial("Zap/1-1", "SIP/405|20|t") in new stack^M -- Called 405Jun 9 06:30:42 WARNING[1226204480]: channel.c:1858 ast_channel_make_compatible: No path to translate from SIP/405-db6d(256) to Zap/1-1(72)Jun 9 06:30:42 WARNING[1226204480]: chan_sip.c:1322 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256)Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJ
un
9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice^M == No one is available to answer at this time^M -- Executing VoiceMail2("Zap/1-1", "u4222760405") in new stack^M -- Playing 'vm-theperson'
(language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/7' (language 'en')^M -- Playing 'digits/6' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/5' (language 'en')^M -- Playing 'vm-isunavail' (language 'en')"
I dont know whatthe message "wait for answer: Unable to forward voice" does mean?. Every time that a call is trying to incoming appears the same log blockshown above.
I dont know what the problem is. Any help may be useful.
Thanks for your help.
Carlos Andres Medina
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[Asterisk-Users] Asterisk on a dual processor machine
Hi there, i have a problem installing asterisk on a dual processor machine.I have Red Hat 9.0 with kernel-smp-2.4.20-6. I did the installation process with no problem, i used the asterisk stable version 1.0. The problem is that the machine has some troubles after Asterisk goes up, the CPU performance goes to 99%, and its all consumed by the asterisk process. I dont know if there is a special process to compile asterisk using dual-processor or maybe a special version or what steps do i have to follow to make asterisk works fine on that machine. The only message error that i have, is when i tried to load the card module...when i put "modprobe wct4xxp" it shows me the following message: "NMI received. Dazed and confused, but trying to continue. You probably have a hardware problem with your RAM chips". After that i follow the rest of loading steps and asterisk goes up just fine. But how i mentionedabove aftera few minutes the CPU performance goes to 99% and the machine is impossible to handle. Thanks for your time. Carlos Andres Medina CVCOL S.A Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
