Re: [Asterisk-Users] Sangoma A104 - D-Channel problem
Hi, Did you tried to set your DMA or SATA as described at message Sangoma A102 cards testing FIXED? Daniel Kumak wrote: Hello, I have following problem with Sangoma A104 card: CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Any idea how to fix it? My configs: zaptel.conf: span=1,1,1,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf [channels] switchtype=euroisdn pridialplan=unknown overlapdial=no usecallerid=yes hidecallerid=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no jitterbuffers=4 echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=yes rxgain=0.5 txgain=0.9 immediate=no amaflags=billing adsi=no busydetect=no callprogress=no context = zapline1 switchtype = euroisdn group=1 signalling = pri_cpe channel = 1-15 channel = 17-31 cat /proc/zaptel/1 : Span 1: WPE1/0 wanpipe1 card 0 HDB3//CRC4 1 WPE1/0/1 Clear (In use) 2 WPE1/0/2 Clear (In use) 3 WPE1/0/3 Clear (In use) 4 WPE1/0/4 Clear (In use) 5 WPE1/0/5 Clear (In use) 6 WPE1/0/6 Clear (In use) 7 WPE1/0/7 Clear (In use) 8 WPE1/0/8 Clear (In use) 9 WPE1/0/9 Clear (In use) 10 WPE1/0/10 Clear (In use) 11 WPE1/0/11 Clear (In use) 12 WPE1/0/12 Clear (In use) 13 WPE1/0/13 Clear (In use) 14 WPE1/0/14 Clear (In use) 15 WPE1/0/15 Clear (In use) 16 WPE1/0/16 HDLCFCS (In use) 17 WPE1/0/17 Clear (In use) 18 WPE1/0/18 Clear (In use) 19 WPE1/0/19 Clear (In use) 20 WPE1/0/20 Clear (In use) 21 WPE1/0/21 Clear (In use) 22 WPE1/0/22 Clear (In use) 23 WPE1/0/23 Clear (In use) 24 WPE1/0/24 Clear (In use) 25 WPE1/0/25 Clear (In use) 26 WPE1/0/26 Clear (In use) 27 WPE1/0/27 Clear (In use) 28 WPE1/0/28 Clear (In use) 29 WPE1/0/29 Clear (In use) 30 WPE1/0/30 Clear (In use) 31 WPE1/0/31 Clear (In use) Information from dmesg: Processing WAN device wanpipe1... wanpipe1: Locating: A104 card, CPU A, PciSlot=8, PciBus=0 wanpipe1: Found: A104 card, CPU A, PciSlot=8, PciBus=0, Port=0 PCI: Found IRQ 10 for device 00:08.0 wanpipe1: AFT PCI memory at 0xEE00 wanpipe1: IRQ 10 allocated to the AFT PCI card wanpipe1: Initializing for SMP wanpipe1: Starting AFT Quad Hardware Init. wanpipe1: Enabling front end link monitor wanpipe1: Global Chip Configuration: used=1 wanpipe1: Global Front End Configuraton! wanpipe1: T1/E1/J1 Global configuration! wanpipe1: AFT Data Mux Bit Map: 0x76543210 wanpipe1: Setting E1 configuration (Port 1)! wanpipe1: All channels enabled wanpipe1: Front end successful wanpipe1: AFT Security: UnChannelised wanpipe1: Configuring Device :wanpipe1 FrmVr=8 wanpipe1:Global MTU = 1500 wanpipe1:Global MRU = 1500 wanpipe1:Data Mux Map = 0x76543210 wanpipe1: Configuring Interface: w1g1 wanpipe1:w1g1: Running in TDM Voice mode. wanpipe1: AFT Fifo Level Map: 0x02082082 wanpipe1: Registering interface to Zaptel span # 1! wanpipe1:MRU :248 wanpipe1:MTU :248 wanpipe1:HDLC Eng :Off (Transparent) wanpipe1:Data Mux Ctrl :On wanpipe1:w1g1: Active channels = 0xFFFE wanpipe1:w1g1: Setting first time slot to 1 wanpipe1:w1g1: Config for Transparent mode: Idle=0 Len=248 wanpipe1:w1g1: Allocating 65 dma skb len=256 Chaining=Off ___ 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 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] SIP termination in Brazil
Hi Han, Our company can offer you a SIP termination in Brazil up and running. Daniel Johannes van Hulst wrote: Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han ___ 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 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] SIP termination in Brazil
Olá Julio, Também oferecemos IAX2. Daniel Julio Arruda wrote: Daniel Bichara wrote: Hi Han, Our company can offer you a SIP termination in Brazil up and running. Daniel IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio de Janeiro. Johannes van Hulst wrote: Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? ___ 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 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] H323 outgoing calls
Hi Darren, Ok, asterisk's H.323 channels works fine. Do you know why you get disconnect from (or can't connect to) your provider? There are some debug available (h.323 debug command). Probably, if you must use G.729 or G.723, you should know you need to buy licenses for this codecs. If you wish to use G.729, you can buy your licenses from Digium, install the licenses and recompile * with G.729 support. Daniel Darren Wiebe wrote: Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider has not been able to help a lot but they have confirmed that I am using the correct dial string. They have also confirmed that they only support G729 and G723. Sorry about the long post, I thought I better send all the info though. Darren Wiebe [EMAIL PROTECTED] chan_h323.so The NuFone Network's Open H.323 Channel 0 This is the result of the following callfile: -- Attempting call on h323/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Aug 19 09:18:30 WARNING[-1273607248]: channel.c:1659 ast_request: No translator path exists for channel type h323 (native 257) to 64 Aug 19 09:18:30 NOTICE[-1273607248]: channel.c:1597 __ast_request_and_dial: Unable to request channel h323/[EMAIL PROTECTED] Aug 19 09:18:30 NOTICE[-1273607248]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 0 Callfile: MaxRetries: 2 extension: Channel: h323/[EMAIL PROTECTED] CallerID: LAKEVIEW 4037422000 This is my h323.conf file: ; The NuFone Network's ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g723.1; Hm... Proprietary, don't use it... allow=g729 dtmfmode=rfc2833 gatekeeper = 65.17.207.253 context=incoming [h323] type=h323 prefix=1010 context=outgoing If I enable all the codecs and switch debug on then this is my output. -- Attempting call on h323/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Aug 19 09:23:09 NOTICE[-1273545808]: pbx_spool.c:235 attempt_thread: Call failed to go through, reason 1 ___ 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 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] Pulse dialed digit recognization
Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a pulse dial telephony. Could * detect digits dialed on pulse telephones? Daniel ___ 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] Channel Bank
You can use VoiceTronix boards. Joe Pukepail wrote: Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ 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 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] VOIP CBQ BandLimit HELP!!
Hi Carlos, Try HTB. It is better than CBQ, requires less CPU and have a better help: http://luxik.cdi.cz/~devik/qos/htb/ Daniel Carlos Arnt wrote: Hi all, Reading about CBQ on internet i can say "I dont understand well" ;) So anyone that has a good background can help me out with this simple question ? I just want priorize my UDP packets to always has 90% of my link when use a VOIP connection with asterisk. My asterisk run in the same machine then my firewall. How then can i : 1 - Mark the packets with iptables then i will know TCP and UDP packets then come in and out 2 - Use CBQ to put a prio=1 in the UDP Packets then i will always know that when a VOIP conn start will always have the best rate of my link. I think i know how mark the packets with the Iptables. iptables -t mangle -A PREROUTING -p tcp -j MARK --set-mark 9000 iptables -t mangle -A PREROUTING -p udp -j MARK --set-mark 9002 and iptables -t mangle -A OUTPUT -p tcp -j MARK --set-mark 9001 iptables -t mangle -A OUTPUT -p udp -j MARK --set-mark 9003 I think that i mark all UDP and TCP packets. So i just need use a CBQ RUle (Now it's the worst) Honestly i dont know .. So let's see. DEVICE=eth0,10Mbit,1Mbit RATE=112Kbit WEIGHT=1Kbit MARK=9000 etc etc I use an 256kbits(Down) - 128Kbits(Up) ADSL connection Then i have PPP0 and my eth1 for my internet net. Just need put the best priority to all UDP Packets forcing the rest of services like SMTP/POP3./HTTP etc that use TCP in the low priority Can anyone help me ? Because i think my Voip has a poor quality because this (Heavy use of mail and http services). Thanks alot for helping out. Carlos I ___ 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 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] VOIP CBQ BandLimit HELP!!
joachim wrote: Daniel, Do you have a working firewall ruleset for HTB, optimized for voip ? No but you can build your own following htb tutorial. Daniel Joachim. (Zoa) At 10:55 1/06/2004, you wrote: Hi Carlos, Try HTB. It is better than CBQ, requires less CPU and have a better help: http://luxik.cdi.cz/~devik/qos/htb/ Daniel Carlos Arnt wrote: Hi all, Reading about CBQ on internet i can say "I dont understand well" ;) So anyone that has a good background can help me out with this simple question ? I just want priorize my UDP packets to always has 90% of my link when use a VOIP connection with asterisk. My asterisk run in the same machine then my firewall. How then can i : 1 - Mark the packets with iptables then i will know TCP and UDP packets then come in and out 2 - Use CBQ to put a prio=1 in the UDP Packets then i will always know that when a VOIP conn start will always have the best rate of my link. I think i know how mark the packets with the Iptables. iptables -t mangle -A PREROUTING -p tcp -j MARK --set-mark 9000 iptables -t mangle -A PREROUTING -p udp -j MARK --set-mark 9002 and iptables -t mangle -A OUTPUT -p tcp -j MARK --set-mark 9001 iptables -t mangle -A OUTPUT -p udp -j MARK --set-mark 9003 I think that i mark all UDP and TCP packets. So i just need use a CBQ RUle (Now it's the worst) Honestly i dont know .. So let's see. DEVICE=eth0,10Mbit,1Mbit RATE=112Kbit WEIGHT=1Kbit MARK=9000 etc etc I use an 256kbits(Down) - 128Kbits(Up) ADSL connection Then i have PPP0 and my eth1 for my internet net. Just need put the best priority to all UDP Packets forcing the rest of services like SMTP/POP3./HTTP etc that use TCP in the low priority Can anyone help me ? Because i think my Voip has a poor quality because this (Heavy use of mail and http services). Thanks alot for helping out. Carlos I ___ 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 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 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] Asterisk on Compact PCI platform
Hi Jay, I am working on this. I am using a 256MB CF. I will keep you informed. Daniel Jay Milk wrote: Since this is related... Does anyone have Asterisk working on a Flash-drive? I was considering this as an alternative to having a harddrive in my machine, thus keeping down noise and heat. A 512MB CF card should be plenty to get Linux and * booted, another 64 or 128MB card should be plenty for voice-mail and such. Any takers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gelson Dias Santos Sent: Wednesday, May 19, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk on Compact PCI platform David H Hickman wrote: I have it working on an industrial single board pc. :) Could you post some more info about your setup? Like board brand/model, what kind of interfaces are you using and even some photos :-) Seems a very interesting project... is there anybody else running a small/compact asterisk system? I would love to have such a small system that I could send to parents, instruct them to turn it on and plug their pstn line and broadband connection and have a pstn x sip intelligent call router that requires no user intervention. Gelson David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 18, 2004, at 8:42 PM, Jacques Leisy wrote: Anybody running * on a compact PCI platform? I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000? Any hope to ever get * running on that platform? Linux Suse 9.0 is running fine Thanks Jacques ___ 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 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 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] asterisk with E1
Hi Thomas, Check your /etc/zaptel.conf and /etc/asterisk/zapata.conf. Probably you have not configured your channels at zapata.conf. Daniel Thomas Schroeter wrote: Hi, I use asterisk with a Digium E1 (wct1xxp). On my old server, everything went fine, but after having built the card to a new one, I only have problems: -- Executing Dial([EMAIL PROTECTED]:18308]/1, Zap/1/853) in new stack May 15 14:10:37 NOTICE[15376]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time ztcfg shows no errors. So where's my problem?!? Regards, thomas PS: zapata.conf: [...] switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15,17-31 zaptel.conf: span=1,0,0,ccs,hdb3,crc4 #,yellow bchan=1-15,17-31 dchan=16 ___ 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 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] Ring Tone - SIP / IAX2
Hi, I am running * CVS 2004-04-05 version. I am having problems to receive ring tone when a SIP device connect to my * box and this box connect another * and then to PSTN: SIP PHONE --- * A --- IAX/2 -- *B E1/PRI/ DIGIUM - PSTN I did not have this problem when I was running version 7.1 on all boxes. I followed threads at list about ring tone problem when connecting SIP devices. Someone told there are problems to receive ring tone when connected to SIP starting at version Stable_1. Is it a bug? Is there a patch? Should I downgrade all my boxes to v.7.1? Thanks in advance, Daniel ___ 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] asterisk/oh323 segfaults
I have the same problem but I am not running RedHat. Daniel Chris Wik wrote: included in this email is a backtrace of a crash on an incoming h.323 call, and also my /etc/asterisk/oh323.conf thanks --- /etc/asterisk/oh323.conf --- ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; ;bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.1.2 gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voip-h323 ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; context=more-stuff alias=664 gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 --- backtrace on core file --- # gdb asterisk core.26437 GNU gdb 5.3 Copyright 2002 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i686-pc-linux-gnu... Core was generated by
[Asterisk-Users] Unable to process inband DTMF
Hi All, Since I updated my * (CVS 2004-03-24), daily, I am getting a strange message just before a segmentation fault: Unable to process inband DTMF on 2 frames. What could it be? Should it cause seg.faults? Daniel ___ 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] Upgraded to latest CVS, now no IAX1?
Brian Cuthie wrote: Tor Houghton wrote: On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ 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 Probably a port collision on your NAT box. I believe that IAX and IAX2 use different ports. Or you can deactivate transfers at iax.conf: notransfer=yes Daniel -brian ___ 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 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] 'Busy tone' after hangup
You can insert a PlayTone(busy) at extension.conf to emulate this behavior. Daniel NetOne Administrator wrote: As you see, * generates no busy tone, it hangs up the channel. It's your client which generates the tone. This is not something to be done from *. Regards, Doichin Dokov Ryan Courtnage wrote: Hello, I find that when 2 extensions are connected, and one of the extensions hangs up on the call, the other will receive a busy signal (as if to indicate that the call is over). Does this sound like a config problem, or is it the default behavior of *? Example: [ext-testing] exten = 111,1,Dial(SIP/2001) exten = 111,2,Hangup exten = h,1,Hangup Zap/2 dials 111: -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, SIP/2001) in new stack -- Called 2001 -- SIP/2001-b164 is ringing -- SIP/2001-b164 answered Zap/2-1 After SIP/2001 hangs up: -- Executing Hangup(Zap/2-1, ) in new stack -- Hungup 'Zap/2-1' ... followed by Zap/2 getting a beep-beep-beep-... 'busy tone'. If this is the default behavior, can it be changed? After the remote end hangs-up on a call, I'd expect to hear either dialtone or silence. Thanks Ryan ___ 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 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 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] IAX2 transfers - it's great!!!!
Senad Jordanovic wrote: And yes, there's a config in iax.conf so you can turn it off if you for some reason want to bother B with staying in the middle of the call. Yap. Great stuff :) Just so everyone knows the config is: notransfer=yes It would be good to know what happens with cdr records and call control? The intermediate IAX exits and register CDR. Since it transfers the call, it does not know anything else about that call. ___ 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 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] Asterisk and Speex
WipeOut wrote: Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. Sorry but I disagree. I am using SPEEX and voice quality is much better than GSM and it consumes less bandwidth. Using Speex and Linux is just a make; make install. Although, we MUST encourage OpenSource initiatives or we will pay Licenses forever. Take a look at G.723 or G.729, for example. ITU released this protocols many years ago and we still have to pay royalties. About this kind of license there is another thread talking about... Daniel ___ 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 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] Qualify statement
Hi, Senad Jordanovic wrote: Does anyone know if qualify=XXX should be used ONLY for user agents behind NAT. No, you can use it if you want to monitor the agent. I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP, and * goes to segmentation fault every time it starts. If it is meant to be used just behind NAT fine, but what and how does * monitor user agent status? Ta SJ ___ 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 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] Fax termination in Asterisk
Hi, You can build a solution with spandsp library. You will need an email server too. http://www.opencall.org/instruction Daniel Tomica Crnek wrote: Hi everyone, Is there an application in Asterisk which can be used as a fax receiver? something like: exten = 1234,1,ReceiveFax(...) exten = 1234,2,ForwardReceivedFax( emailaddress ) Tomica ___ 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] Q931 Message - Connect - Billing
Hi Martin,
I don't think I have any playback of answers in my extension. Please,
check the following exten.conf:
[default]
exten = _X.,1,SetVar(VCOL=20)
exten = _X.,2,SetVar(VPRL=0)
exten = _X.,3,SetVar(VDIG=0${EXTEN})
exten = _X.,4,SetCIDNum(123456|a)
exten = _X.,5,Wait(1)
exten = _X.,6,Goto(dial,s,1)
[dial]
exten = s,1,ResetCDR()
exten = s,2,Dial(IAX2/[EMAIL PROTECTED]/${DIG})
exten = s,3,Busy
Thanks in advance,
Daniel
Martin Pycko wrote:
If you have playback first before dialing the right extension on the other
side then that's why you have the call answered right away ...
Also voicemail answers and many other applications.
Martin
On Tue, 16 Mar 2004, Daniel Bichara wrote:
Hi All,
I have posted before asking for a Connect message sent from Zap
(ISDN/PRI - by *) when receiving a call (incoming) and dialing to
another extension. To clarify the situation, I will describe the problem:
1) My * box is connect to a Cisco (E1-ISDN/PRI) using a crosscable.
2) Sending a call (outbound) to Cisco, I receive Q931 "Connect (15)"
message from Cisco only after the other side answer.
3) Receiving a call (inbound) from Cisco, I call an extension at another
* connected via IAX2:
X.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Before the other side (${EXTEN} at 192.168.110.2) answer, * sends a Q931 "Connect (15)" message to Cisco and it starts billing the call.
Cisco bills even if no one answer the other side or if its busy.
I tried to setup "overlapdial=yes" at Zapata.conf but E1 disconnects by "Timeout" (T_313 expires - 4secs) before IAX2 complete the call.
I noticed I get a message: "Progress Description: Called equipament is non-ISDN" from "pri debug span". Is that correct?
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
Message type: CONNECT (7)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan:
0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called equipment i
s non-ISDN. (2) ]
-- IAX2[192.168.110.2:4569]/1 stopped sounds
-- IAX2[192.168.110.2:4569]/1 is ringing
Any clue? Thanks in advance.
Daniel Bichara
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Re: [Asterisk-Users] NuFone?
Hi All, We know everyone can offer services. May we build a interconnected * network all over the world to offer best conditions each other? We can set a service level agreement and try ;-) Any one? Daniel [EMAIL PROTECTED] wrote: Since everyone is offering their services then: USA - £0.016 (~ 2.9c) UK - £0.016 (~ 2.9c) Europe - £0.02 (~ 3.6c) UK 0800 - FREE SIP / IAX termination. auto-provisioning, web-based billing, call history, on-line top-up, credit-card payments. Not US-based though :-( Tan www.voiptalk.org www.iaxtalk.co.uk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer Sent: 17 March 2004 19:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NuFone? Doug Harris wrote: Hi, Seems like there arn't any alternative to NuFone either ? Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached. Doug If you want SIP/IAX termination from someone other than NuFone for the same price, you can contact me. We can offer that. John ___ 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 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 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] Transparent Switch - PRI / IAX2 / PRI
Hi,
I wish * switch calls "transparent" from one port PRI to another * using IAX. If I have a line at extension.conf like this:
_X.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I get a connection PRI Q931 message before ringing other side:
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
Message type: CONNECT (7)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan:
0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 1 ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called equipment i
s non-ISDN. (2) ]
-- IAX2[192.168.110.2:4569]/1 stopped sounds
-- IAX2[192.168.110.2:4569]/1 is ringing
How could I deal with this signalization? I wish to receive CONNECT after Answered.
-- Zap/1-1 answered IAX2[[EMAIL PROTECTED]:4569]/1
Thanks in advance.
Daniel
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[Asterisk-Users] Native Bridge and Billing
Hi all, I am connecting two * (A and B) using a third * (C) as passthru and billing control. All connections are IAX-2. So, when A wants to call someone outside, it Dials to C. C analyzes the extension number and redirects it to the appropriate destination at B, billing the call: A (exten 223) calls extension 978 at C C knows extension 978 is B extension 10978 and calls it - B accepts the call to 10978 from C When connection between C and B is estabilished, C starts native bridge mode, transfering call control. For C, call ended and it bills as it longs only few seconds. Should I disable native bridge? How? I need C bills the call and controls it. Thanks in advance, Daniel ___ 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] Transfer and Native Bridge unwanted - was Native Bridge and Billing
Hi all, Ok. Now I know I can't bill a call when I have a native bridge betweens *. And I do not want a Native Bridge. How could I disable native bridge? I tried notransfer=yes but connection tries to start a native bridge and then closes. Daniel ___ 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] E1 - Signal - PBX connection
Hi all,
I've posted another message yesterday about the same problem. I will try
to be more detailed to get some help from the list:
I am connecting two PBX (PBX-A and PBX-B) using two * (*-A and *-B).
Asterisks are connected via IAX2. The PBXs area connected to each *
using an E1.
PBX-A -- E1 -- *-A -- IAX2 -- *-B -- E1 -- PBX-B
When an extension from PBX-A calls an extension at PBX-B, PBX-A calls
*-A that calls *-B and then PBX-B. The problem is: if extension at B is
busy, *-A returns Normal Call Clear to PBX-A and PBX-A bills the call
normally. I contacted PBX support and they said me *-A should return a
different value to PBX-A (I think it is busy detected).
#
iax.conf at PBX-A:
[pbxb]
type=friend
auth=md5
secret=secret
bandwidth=low
disallow=all
allow=speex
#
extensions.conf at PBX-A:
[default]
exten = _.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I tried to insert |r for ringback-only at Dial command and there is no
difference.
I search the mailing list and I found some emails about Call Signalling
and IAX protocol.
Any clue?
Daniel
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[Asterisk-Users] IAX and E1 Call State
Hi all, I am connecting two PBX using two * and IAX2. There is one E1 connected to each *. I receive a call from PBX-A and dial to *-B / Zap-g1 (PBX-B). If the destination is busy or ring until I put on hook, *-A returns normal Call Clear and the PBX-A (attached to *-A) bills the call. PBX-A -- E1 -- *-A -- IAX2 -- *-B -- E1 -- PBX-B I search the mailing list and I found some emails about Call Signalling and IAX protocol. Could IAX return the correct Call State to E1 attached to A? Daniel ___ 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] Brazilian Protocol
Hi Alex, Alex G Robertson wrote: Hi all, I would like to have some information about your TE410p and TE405p cards compatibility with telephony protocols adopted in Brazil. - When in E1 mode, does it support R2 DIGITAL MFC 5C ? You need a R2 converter. R2lib is under construction. - When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ? - Is asterisk pabx enough to implement these protocols o we need other drivers/software? - Do you know if any brazilian telco (Telemar, Vesper, Embratel, GVT, Telefonica, BRT, etc) supports custumers with you hardware? GVT supports E1/ISDN. Daniel Can you help me? Thanks in advance ___ 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] Brazilian Protocol
Marcio, Marcio Gomes wrote: You need a R2 converter. R2lib is under construction. - When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ? - Is asterisk pabx enough to implement these protocols o we need other drivers/software? - Do you know if any brazilian telco (Telemar, Vesper, Embratel, GVT, Telefonica, BRT, etc) supports custumers with you hardware? GVT supports E1/ISDN. What is the ISDN Sinalization from GVT, I think using PRI Lines with telco in Brazil, we will have comaptibility with Digium Borads! Is it correct ? A GVT oferece linhas E1/ISDN-PRI. Historicamente, o Brasil adotou a sinalização R2Digital como padrão para circuitos E1 e a maioria das Operadoras não oferecem outra sinalização. Na minha experiência: a BrT, Embratel e Telemar oferecem apenas R2Digital. A Telefônica, sob consulta. Daniel Best Regards, []s Marcio Gomes Daniel Can you help me? Thanks in advance ___ 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 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 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] R2lib
Marcio, Marcio Gomes wrote: Hello All, I forgot this question in my last post .. Where is the primary site to R2Lib ? There is no primary site. A scratch have been release last year but its developer said to me this source is a junk. He will release a new beta version in the future. Daniel Best Regards, Marcio Gomes Marcio Gomes wrote: You need a R2 converter. R2lib is under construction. - When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ? - Is asterisk pabx enough to implement these protocols o we need other drivers/software? - Do you know if any brazilian telco (Telemar, Vesper, Embratel, GVT, Telefonica, BRT, etc) supports custumers with you hardware? GVT supports E1/ISDN. What is the ISDN Sinalization from GVT, I think using PRI Lines with telco in Brazil, we will have comaptibility with Digium Borads! Is it correct ? Best Regards, []s Marcio Gomes Daniel Can you help me? Thanks in advance ___ 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 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 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 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] Codec Order / Preference
Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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] Codec Order / Preference
Regovich, Timothy wrote: Really? Did you try disallow=all Allow=speex Allow=gsm Allow=alaw Yes and it did no work. ? T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 23, 2004 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codec Order / Preference You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 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] Voicepulse Connection
Hi, I have two * connecteds and I wish a phone connected to * #1 calls PSTN via Voicepulse connected to * #2, as follows: telephone --- Asterisk #1 Asterisk #2 Voicepulse When I dial 81-90... (japan), * #1 will route call to Voicepulse at *#2. Everything works between #1 and #2 but when #2 calls Voicepulse I get an error message: -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call rejected by 66.234.228.132: No such context/extension I am clueless!!! What could it be? Follow my confs... Exten.conf - *#1 exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) # Exten.conf - *#2 [outvoicepulse] exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _.,2,Congestion # Iax.conf - *#2 [voicepulse] context=VPWS secret=password auth=md5 type=friend host=66.234.228.132 disallow=all allow=speex allow=gsm jitterbuffer=no Daniel ___ 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] Connection Problem - GrandStream
Hi, I am connecting two * and a GS phone. I can call Zap/g1 at *#1 from *#2 Zap/1. It works. But when I try to call Zap/g1 at #1 from a GS (SIP) phone connected to #2, I get an error message at #1 that looks like * #1 does not know which codec I use to connect my SIP phone: Feb 21 10:36:16 NOTICE[278545]: channel.c:1448 ast_set_write_format: Unable to find a path from UNKN to GSM Diagram: This works: Zap/1 at #2 calls #1 Zap/g1 Zap/1 -- Asterisk #2 --- Asterisk #1 -- Zap/g1 Here I have a problem: GS(SIP) -- Asterisk #2 -- Asterisk #1 -- Zap/g1 Daniel ___ 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] IAX Connection - Voicepulse
Hi, I am call Japan via Voicepulse. My IAX Connection to Voicepulse was sucessfull. But when I put a call (dial), I get an error message: Feb 19 22:12:23 WARNING[81926]: chan_iax2.c:1128 attempt_transmit: Max retries exceeded to host 66.234.228.132 on IAX2[voicepulse]/4 (type = 6, subclass = 1, ts=1, seqno=0) My extension.conf: exten = s,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) What could be? Daniel ___ 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] PRI error or what?
C. Maj wrote: On Tue, 17 Feb 2004, Tomica Crnek waxed: I have TE410P with two E1 links connected. It is working ok, but suddenly, from time to time I got this and it goes on and on for a few minutes during which period I can't establish new calls == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up This could be a problem with your telco. I had the same thing for a couple of months before they finally identified it as their "bad cable pair" in outside wiring. Run a trace on the PRI from *, and find someone at the telco to do the same on their end. I solved this problem when I changed my sync source: span = 1,1,0,ccs,hdb3 Daniel Good luck, --Chris ___ 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] Termination - Cuba
Hi, I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High traffic. Thanks in advance, Daniel ___ 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] Termination - Cuba
Amaury Jacquot wrote: Daniel Bichara wrote: Hi, I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High traffic. the only one you'll get on the phone there is Fidel Castro (which is the only one to have internet access too) :D You are right! But someone must have a rate better than USD$0.90 / minute. Daniel Amaury Thanks in advance, Daniel ___ 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 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 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] Unable to create vpb channel
You need to insmod vpb module. Daniel Steven Kawuma wrote: Hi, My vpb.conf now reads: [interfaces] echocancel = on board = 1 context = parlix_agents ; Note that V6PCI channel numbers start at 7! mode = fxo channel = 7 channel = 8 mode = dialtone channel = 9 channel = 10 channel = 11 channel = 12 But I still get the same error. Just in case, `lsmod | grep vpb` gives vpb 135264 0 (unused) vpbhp 224128 1 Thanks in advance. Steven. On Mon, 2004-02-09 at 10:24, Steven Kawuma wrote: Hi all, I'm using a voicetronix openswitch6 card with asterisk. When I try to dial the vpb phone from my application, I get t he following error: -- Executing Dial(Zap/1-1, vpb/1-9|10|mtT||/usr/local/sbin/parlix_dial_event 2 196) in new stack -- 1-9 requested, got: [None] NOTICE[524311]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'vpb' What does it mean? Below is my vpb.conf: [interfaces] echocancel = on board = 1 context = agents ; Note that V6PCI channel numbers start at 7! mode = fxo channel = 7 ;channel = 8 mode = dialtone ;channel = 11 ;channel = 12 ___ 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] looking for iax termination
Hi, We have termination based on IAX and SIP at Brazil. Daniel [EMAIL PROTECTED] wrote: Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. ___ 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] ETSI PRI ISDN Signalling
Thanks. I solved this problem using a cross-cable. Daniel CW_ASN - Gus wrote: Please send your zaptel.conf to see what's going on. - Original Message - From: "Daniel Bichara" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 4:38 PM Subject: [Asterisk-Users] ETSI PRI ISDN Signalling Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel ___ 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 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 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] ETSI PRI ISDN Signalling
Samuel Jimenez wrote: Hi, Assuming that the problem *is not* soft settings I would recommend you to verify that the channel mapping is the same in your adapter as in your E100P. Some times they do not match, and the D channel does not arrive where expected at each end. Hi Sam, My problem was a cross-cable. Now it is ok. Thanks Daniel As you said, just a clue. Rgds Sam\\\ - Original Message - From: Daniel Bichara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 1:38 PM Subject: [Asterisk-Users] ETSI PRI ISDN Signalling Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel ___ 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 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 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] ETSI PRI ISDN Signalling
Hi All, I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue? Daniel ___ 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] R2 support
CW_ASN wrote: CW_ASN - Gus wrote: Ok, it's old and clunky, but in some countries like Brazil, Argentina and China is the only alternative. Only alternative??? Why is the only alternative? All mayor carriers in Argentina and Brasil have PRI signalling, at the same price. Sorry, but you are wrong. I am from Brazil and E1-ISDN is not avaible all over the country. Daniel Maybe, you don't have big carriers in all country... Maybe Telefonica (the same from .ar) is not big enough! ___ 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] R2 support
CW_ASN - Gus wrote: Ok, it's old and clunky, but in some countries like Brazil, Argentina and China is the only alternative. Only alternative??? Why is the only alternative? All mayor carriers in Argentina and Brasil have PRI signalling, at the same price. Sorry, but you are wrong. I am from Brazil and E1-ISDN is not avaible all over the country. Daniel ___ 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 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] Voicetronix OpenLine4: disable answering on a particular channel delay before dial
Hi Terence, Terence Parker wrote: Hi there, After a lot of valuable insights from the list, incoming and outgoing calls finally work through OpenLine4! Thanks for all the input! Now I have 2 minor issues: Sometimes Voicetronix dials too quickly before an actual dial tone is obtained from the phone company. E.g. Voicetronix picks up a line and then dials immediately, whereas actually it took the phone company may be half a second to actually make the line available to gave a dialtone. As a result? 90% of the time, the first digit dialed was not received by the phone company. Is it possible to tell voicetronix to wait a second or two before dialing? Try to insert a comma "," before the number you dial. Secondly, I have a phone line plugged into channel 2 that I don't want Asterisk to answer. I only want ASterisk to use it to dialout. So I need to configure Asterisk somehow to ignore incoming calls on channel 2. Is this possible? in /etc/asterisk/vpb.conf , before "channel = 2" insert a new context definition and config extensions.conf to ignore dial in: vpb.conf: context = default channel = 1 context = nodialin channel = 2 in extensions.conf, insert: [nodialin] exten = s,1,Congestion Daniel Thanks! Terence ___ 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 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] Asterisk drops calls - E100P
Don Pobanz wrote: On Wednesday, January 14, 2004 5:48 AM, Daniel Bichara [SMTP:[EMAIL PROTECTED]] wrote: Hi, Once a day, * drops all calls (E100P board). Yesterday, I updated * version to CVS but I got the problem again today. Monitoring log files, I found this messages just before: Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Short write: -1/5 (Unknown error 500) Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Few minutes after this, everything becomes fine. Any clue? just a guess here... The simple answer is have you verified that loop timing is set up in zaptel.conf. If not in loop timing a slip could cause the drop. Loop timing on span 2 as primary timing would be: span=2,1,0,esf,b8zs Does this happen at the same time every day? If so it does not sound like a timing issue. If at random times, it could be. Hi Don, You are right! Yesterday, my telco called me about slip. I changed timing and now it is ok. Daniel Daniel Don Pobanz ___ 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 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] Asterisk drops calls - E100P
Hi, Once a day, * drops all calls (E100P board). Yesterday, I updated * version to CVS but I got the problem again today. Monitoring log files, I found this messages just before: Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Short write: -1/5 (Unknown error 500) Jan 14 09:14:15 WARNING[81926]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 25 failed: Unknown error 500 Few minutes after this, everything becomes fine. Any clue? Daniel ___ 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] Best Linux Distribution
[EMAIL PROTECTED] wrote: Hi my question is: which is the best distribution to work with asterisk? Hi Mark, I am working on a distro called SAX built to optimize * and routing. It works with RPMs and its HFS is RedHat like. I built all packages by hand and created RPMs packages. It is in beta version by now. More few days and I will release an ISO image. Daniel thanks mark ___ 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 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] R2 Digital - Brazil
Hi Steve, I got libr2 from CVS and I am trying to make it run. First, I found one error at chan_zap.c (calling zt_new with wrong parameter). Where could I find the signalling differences between countries ( I am trying to run .ar at .br)? Daniel Steve Underwood wrote: Hi Daniel, You will find libr2 is only about 10% of an implementation, and a bad one at that. I now have 95% of a good implementation, but its not yet released. Regards, Steve Daniel Bichara wrote: Hi all, I will start testing libr2 for brazilian R2. Any clue? Daniel ___ 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 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] R2 Digital - Brazil
Hi Steve, Yes, I did. But I wish I could help. Daniel Steve Underwood wrote: Hi Daniel, Did you read what I wrote? libr2 is only 10% of an implementation. It doesn't work. Its a useless piece of junk. Forget about it. As for the signalling differences between countries, that is something of a pain to find out. There is not much info about variants of R2 available on the internet. I pieced together the picture I have now, which may not be complete. Regards, Steve Daniel Bichara wrote: Hi Steve, I got libr2 from CVS and I am trying to make it run. First, I found one error at chan_zap.c (calling zt_new with wrong parameter). Where could I find the signalling differences between countries ( I am trying to run .ar at .br)? Daniel Steve Underwood wrote: Hi Daniel, You will find libr2 is only about 10% of an implementation, and a bad one at that. I now have 95% of a good implementation, but its not yet released. Regards, Steve Daniel Bichara wrote: Hi all, I will start testing libr2 for brazilian R2. Any clue? Daniel ___ 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 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] CAS Idle definition bits ?
Hi, Could some one explain what are the 4 bits we should define after cas setup (zapata.com) (CAS Signalling requires idle definition in the form ':' ? Thanks in advance, Daniel ___ 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] E100P - connected to Cisco
Hi, have any one sucessfully connected E100P to Cisco? I am getting singnalling problems (channels becomes up and down)... # Zaptel.Conf: span=2,0,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 # Zapata.conf: callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes group = 2 signalling = pri_net switchtype = national channel = 32-46 channel = 48-62 # My New Cisco Conf: ! Cisco AS5300 - ios c5300-is-mz.122-5.bin isdn switch-type primary-ni isdn voice-call-failure 0 controller E1 3 framing NO-CRC4 pri-group timeslots 1-31 interface Serial3:15 no ip address isdn switch-type primary-ni isdn protocol-emulate network isdn guard-timer 2000 isdn T203 1 isdn T306 3000 isdn T310 6 isdn bchan-number-order ascending no cdp enable My INTENSE debug output: [ [00 [00 01 [00 01 01 [00 01 01 01 [00 01 01 01 ] [00 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter [ [00 [00 01 [00 01 00 [00 01 00 00 [00 01 00 00 08 [00 01 00 00 08 02 [00 01 00 00 08 02 00 [00 01 00 00 08 02 00 00 [00 01 00 00 08 02 00 00 46 [00 01 00 00 08 02 00 00 46 18 [00 01 00 00 08 02 00 00 46 18 03 [00 01 00 00 08 02 00 00 46 18 03 a9 [00 01 00 00 08 02 00 00 46 18 03 a9 83 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80 [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ] [00 01 00 00 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 N(S): 000 0: 0 N(R): 000 P: 0 13 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 28 ] Restart Indentifier: [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- T200 counter expired, What to do... -- Retransmitting 17 bytes [ [00 [00 01 [00 01 00 [00 01 00 01 [00 01 00 01 08 [00 01 00 01 08 02 [00 01 00 01 08 02 00 [00 01 00 01 08 02 00 00 [00 01 00 01 08 02 00 00 46 [00 01 00 01 08 02 00 00 46 18 [00 01 00 01 08 02 00 00 46 18 03 [00 01 00 01 08 02 00 00 46 18 03 a9 [00 01 00 01 08 02 00 00 46 18 03 a9 83 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80 [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ] [00 01 00 01 08 02 00 00 46 18 03 a9 83 9c 79 01 80 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 N(S): 000 0: 0 N(R): 000 P: 1 13 bytes of data -- Rescheduling retransmission (1) [ [00 [00 01 [00 01 01 [00 01 01 03 [00 01 01 03 ] [00 01 01 03 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 001 P/F: 1 0 bytes of data -- ACKing all packets from 0 to (but not including) 1 -- ACKing packet 0, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter [ [02 [02 01 [02 01 00 [02 01 00 02 [02 01 00 02 08 [02 01 00 02 08 02 [02 01 00 02 08 02 80 [02 01 00 02 08 02 80 00 [02 01 00 02 08 02 80 00 4e [02 01 00 02 08 02 80 00 4e 18 [02 01 00 02 08 02 80 00 4e 18 03 [02 01 00 02 08 02 80 00 4e 18 03 a9 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80 [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80 ] [02 01 00 02 08 02 80 00 4e 18 03 a9 83 9c 79 01 80 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000 EA: 1 N(S): 000 0: 0 N(R): 001 P: 0 13 bytes of data -- ACKing all packets from 0 to (but not including) 1 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 32768/0x8000) (Terminator) Message type: RESTART ACKNOWLEDGE (78) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 28 ] Restart Indentifier: [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] Sending Receiver Ready (1) [ [02 [02 01 [02 01 01 [02 01 01 02 [02 01 01 02 ] [02 01 01 02 ]
Re: [Asterisk-Users] E100P - Error 500
Scott Stingel wrote: I get lots of these in a very busy system, along with PRI frame errors/retransmissions. It is my understanding that this is due to an inadequate buffering mechanism in asterisk. Mark Spencer is aware of the problem, and has said he'll work on it soon. In small numbers, these can be safely ignored. Hi Scott, But sometimes it closes or destroys all open Zap channels (put call onhook). Daniel Regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara Sent: Saturday, January 10, 2004 3:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E100P - Error 500 Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:48 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Any clue? Daniel ___ 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 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 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] R2 Digital - Brazil
Hi all, I will start testing libr2 for brazilian R2. Any clue? Daniel ___ 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] E1 - E100P connected to Cisco - problem - HELP
s, Daniel Steve On Wed, 7 Jan 2004, Daniel Bichara wrote: Hi, I am trying to connect * E100P directly to Cisco using an ATM circuit. The ATM circuit is ok (I can connect to Ciscos). I do not understando too much about Cisco syntax, please help me. # Cisco Conf: isdn switch-type primary-net5 isdn voice-call-failure 0 controller E1 3 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial3:15 no ip address isdn switch-type primary-net5 isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn send-alerting no cdp enable # Zaptel.Conf: span=2,1,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 # Zapata.conf: callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes group = 2 signalling = pri_net switchtype = national channel = 32-46 channel = 48-62 Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] E1 - E100P connected to Cisco - problem
7 Jan 2004, Daniel Bichara wrote: Hi, I am trying to connect * E100P directly to Cisco using an ATM circuit. The ATM circuit is ok (I can connect to Ciscos). I do not understando too much about Cisco syntax, please help me. # Cisco Conf: isdn switch-type primary-net5 isdn voice-call-failure 0 controller E1 3 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial3:15 no ip address isdn switch-type primary-net5 isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn send-alerting no cdp enable # Zaptel.Conf: span=2,1,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 # Zapata.conf: callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes group = 2 signalling = pri_net switchtype = national channel = 32-46 channel = 48-62 Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] cant load drivers for TE410P cards
You need to compile HDLC support at kernel. Prefer static then module. Daniel [EMAIL PROTECTED] wrote: hello, I have been using the T1 card with my asterisk for a while now, but an attemp to upgrade the system to use a TE410P card ( using the T1 option) i have a 3.3V motherboard. but when i try to load the module it gives the following errors: #modprobe zaptelO.k # modprobe wct1xxp gives /lib/modules/2.4.18-4/misc/wct1xxp.0:init module:no such device Hint: insmod errors cab caused by incorrect module parameters, including invalid IO and IRQ parameters. /lib/modules/2.4.18-4/misc/wct1xxp:: insmod wct1xxp failed. is there a way of installing the TE410P module? regarsd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 - E100P connected to Cisco - problem
Hi, I am trying to connect * E100P directly to Cisco using an ATM circuit. The ATM circuit is ok (I can connect to Ciscos). I do not understando too much about Cisco syntax, please help me. # Cisco Conf: isdn switch-type primary-net5 isdn voice-call-failure 0 controller E1 3 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial3:15 no ip address isdn switch-type primary-net5 isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn send-alerting no cdp enable # Zaptel.Conf: span=2,1,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 # Zapata.conf: callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes group = 2 signalling = pri_net switchtype = national channel = 32-46 channel = 48-62 Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two E100P boards - could not load zaptel module - Channel 63 - no such device
Hi, I have two E100P boards connected to my PC. I wish to setup two E1-ISDN-PRI lines, the first SPAN with 15 channels and the second one with 30 channels. When I try to load zaptel modules, I get an error message: Loading zaptel framework: Loading zaptel hardware modules: wct1xxp wcusb Running ztcfg: ZT_CHANCONFIG failed on channel 63: No such device Follows my zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 span=2,0,0,ccs,hdb3 bchan=33-47,49-63 dchan=48,64 Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex Codec - Error IAX2
I just installed Speex library and recompiled *. It works fine. Daniel Olle E. Johansson wrote: Brian West wrote: Did you install speex? Its not there by default and you must build extra libs for the codec to work. www.speex.org How does speex integrate into Asterisk? At compile time? I need to document this, was not aware of the need for a third-party library. Any help and explanations welcome! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 API Card Solution
Do I need a special Digium Card (E100-SS7) or use my E100P card and compile the new drivers? Daniel Juan J. Sierralta P. wrote: On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote: Is this useful as a bootstrap for getting SS7 to Asterisk? http://www.sangoma.com/api/p-api-ss7.htm You should check http://www.openss7.org the have an stack and works with an special version of digium cards, dunno if is the same HW with special drivers but it looks much more * friendly. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk crash
Try to run zaptel.init (from zaptel package) Daniel [EMAIL PROTECTED] wrote: Ok what do I do with ztcfg? On Mon, 29 Dec 2003, zoa wrote: try running ztcfg first At 09:26 29/12/2003 -0500, you wrote: Hello all I just checked out the latest zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through the entire make procedures. Everything seemed to go fine however now when I attempt to start asterisk, it says ok but it seems to be immediately crashing. The following messages are displayed in my /var/log/asterisk/messages file for the time right around the crash: Dec 29 10:09:37 WARNING[1074395872]: File chan_zap.c, Line 7341 (setup_zap): Ignoring rxwink Dec 29 10:09:37 ERROR[1074395872]: File chan_zap.c, Line 5159 (mkintf): Unable to get span status: Inappropriate ioctl for device Dec 29 10:09:37 ERROR[1074395872]: File chan_zap.c, Line 7081 (setup_zap): Unable to register channel '1-23' Dec 29 10:09:37 WARNING[1074395872]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Does anyone have any insight on this? Everything compiled fine and was working fine with my previous version of asterisk. Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P - rxgain / txgain
Hi, I am using E100P / ISDN-PRI and voice sound is too loud. May I use negative gains (rxgain, txgain) to make volume lower? What is the range for gains? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 API Card Solution
Steven Critchfield wrote: This is not a flame, but a reminder on why proper quoting is important. Another example of truly bad default behavior by software and why user education is important to fix the Microcrap problem. 3 posts, and not one of the replies used proper quoting. The last didn't even provide a attribute line noting where the quoted material started. I know it has been a while since my last email rant, but this was way over the top for annoying to read. Worst of all, it provided important information that needed to be seen and was on the verge of being ignored due to inefficient communication style. Hi Steven, Here in south Brazil people could say you peed on me! Sorry but I subscribe other lists and I know it is difficult to understand text cadence in some cases. It is not my intention and I will be more carefull next time. Daniel On Mon, 2003-12-29 at 11:39, Ray Burkholder wrote: Current Status: http://www.openss7.org/asterix.html Ray Do I need a special Digium Card (E100-SS7) or use my E100P card and compile the new drivers? Daniel Juan J. Sierralta P. wrote: On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote: Is this useful as a bootstrap for getting SS7 to Asterisk? http://www.sangoma.com/api/p-api-ss7.htm You should check http://www.openss7.org the have an stack and works with an special version of digium cards, dunno if is the same HW with special drivers but it looks much more * friendly. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 API Card Solution
Steve Underwood wrote: That status page tells you the porject has gone nowhere so far. What you need is not a driver. It is a development project! Is there any candidate? Is there a former team project? Could I help? Daniel Regards, Steve Ray Burkholder wrote: Current Status: http://www.openss7.org/asterix.html Ray Do I need a special Digium Card (E100-SS7) or use my E100P card and compile the new drivers? Daniel Juan J. Sierralta P. wrote: On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote: Is this useful as a bootstrap for getting SS7 to Asterisk? http://www.sangoma.com/api/p-api-ss7.htm You should check http://www.openss7.org the have an stack and works with an special version of digium cards, dunno if is the same HW with special drivers but it looks much more * friendly. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex Codec - Error IAX2
Hi, I am trying to connect two * using Speex codec via IAX2. When it starts connection I get an error message : -- Format for call is SPEEX NOTICE[98311]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ALAW to SPEEX NOTICE[98311]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from SPEEX to ALAW Any clue? Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN-PRI - WCT1XXP error
Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 loadzone = us Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZTMonitor - /dev/dsp problem
Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P connected to Cisco
Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig isdn voice-call-failure 0 controller E1 2 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial2:15 no ip address isdn switch-type primary-qsig isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn T310 4 isdn send-alerting no cdp enable voice-port 2:D cptone BR I configured my /etc/zapata.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 Any clue? Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
