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  • Parse (extract) DTMF in Python

    - by Joseph
    If I have a recorded audio file (MP3), is there any way to get the figure out the DTMF tones that were recorded in pure Python? (If pure python is not available, then Java is okay too. The point being that it should be able to run in Google Appengine)

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  • n900 - SIP - Google Voice - DTMF

    - by Walter White
    Hi all, I've been using Google Voice on my n900 and it works reasonably well, but DTMF tones do not work whatsoever. According to this page, it is fixed and has been for quite some time. https://bugs.maemo.org/show_bug.cgi?id=5505 Has anyone had any luck with SIP on the n900 and DTMF tones? An earlier version of Skype on the n900 didn't support DTMF, but that was fixed. I would have thought this bug would have been fixed for some time now. Thanks, Walter

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  • Asterisk does not recognise DTMF tones from mobile phones

    - by Eugene van der Merwe
    We have an Asterisk 1.8.7.0 (the Elastix derivative) switchboard. Every since a month ago, seemingly out of the blue, the switchboard does not recognise DTMF tones any more from mobile phones. Testing the switchboard using 7777 works. Testing the switchboard from a normal phones works. Testing the switchboard from a mobile phone fails. Looking at the log file I can't see anything. I used 'asterisk -rvvvv' and 'tail -f /var/log/asterisk/full' to see the live output and scan the logs. I guess I don't see anything because it's simply not recognising the DTMF tones. I did brief research and found an old setting for SIP phones, 'rfc2833compensate=yes', and tried adding this to 'sip_general_custom.conf'. After that I did 'core restart when convenient' but that didn't make any difference. Could anyone give me some additional troubleshooting steps?

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  • LYNC 2010 Dial-In in Meeting DTMF issue

    - by user140116
    We are facing an issue in the LYNC2010 dial-in to a meeting. We redirect an Asterisk number to LYNC, whitch connects successfully in the dial-in plan of LYNC. After calling from external network to the given number, we hear LYNC aswering and prompting us to enter the PIN and afterwards the hash key. I should mention that all other dials to LYNC from Asterisk and vise versa are routed successfully. Also all DTMF we send to Asterisk from the phone (IVR, Extension, PIN etc) are routed also fine Afterwards we press the appropriate pin folowed by the hash keyand we get 'Sorry I can't find meeting with that number' Some pros mentioned that it might be dtmfmode=RFC2833 or dtmfmode=auto in Asterisk (All checked and tried). Some pros mentioned, that there is a problem in geeral in LYNC and DTMF (even with Cisco Call Manager). Some other pros mentioned that chack box 'Enable refer support' in Voice Routinh\Trunk Configuration' in LYNC has to be unchecked (Also tested). The problem stil remains and there is no way to enter a meeting room by dial-in. ANY idea would be appreciated!!!!!!!!

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  • Programatically Dial a Phone number and pass DTMF using the iPhone SDK

    - by L. DPenha
    How do you programatically do the following from the iPhone SDK 1) Programatically Dial a Phone Number through the iPhone SDK 2) Bypass the dial / cancel prompt that the iPhone brings up 3) Send additional DTMF after the number is dialed just like how you would program pauses into a regular phone. I know you can make a tel:// call but the issue is that it brings up the dial / cancel prompt and after that it prevents any future DTMF from being sent.

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  • Call a Phone number and pass DTMF using the iPhone SDK

    - by datag
    Please suggest me to achieve the below requirement using iphone SDK. 1) Programatically Dial a Phone Number through the iPhone SDK 2) Send additional DTMF after the number is called. After googled found tel URL scheme is available to make a call. Not sure about sending DTMF tones. Please help me. https://developer.apple.com/iphone/library/featuredarticles/iPhoneURLScheme_Reference/Articles/PhoneLinks.html Thanks

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  • Touch tone/DTMF "translator"?

    - by IVR Avenger
    Hi, all. Does anyone know of a device that can be plugged into a telephone handset that will display the value of any telephone keys pressed by someone on the other end? So, if I'm on the phone and the other party dials "1234", I'd look down at the device plugged into my handset, and see "1234." Thanks! IVR Avenger

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  • iPhone 3.5mm jack based application

    - by maverick
    I want to encode data via a DTMF encoder and send it back to the iPhone via the 3.5mm Jack. Is it possible to send data back into the 3.5mm jack. conventionally audio signals are sent out over the iPhone 3.5mm jack? Is there provision to deal with DTMF and 3.5mm jack based input applications in Iphone's External Accessory framework?

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  • How do I activate my gizmo5 phone number in Google Voice? [closed]

    - by Sorin Sbarnea
    I wasn't able to activate my gizmo5 number because Google Voice activation(verification) requires you to enter two dial tones (DTMF) and they did not work at least not with these two variants: Using gizmo5 PC client using fring from Iphone as gizmo5 SIP client Redirecting gizmo5 to a US mobile number None of the above methods worked for me. Any ideas? More info: http://www.google.com/support/forum/p/voice/thread?tid=1d8c1d99721e3509&hl=en http://googlevoices.blogspot.com/2009/04/forwarding-sip-calls-to-google-voice.html

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  • "RFC 2833 RTP Event" Consecutive Events and the E "End" Bit

    - by brian_d
    Hello, I can send out a RFC 2833 dtmf event as outlined at http://www.ietf.org/rfc/rfc2833.txt When I do set the E "End" bit, but leave it as 0, I get the following behaviour: If for example keys 7874556332111111145855885#3 were pressed, then ALL events would be sent and show up in a program like wireshark, however only 87456321458585#3 would sound. So the first key (which I figure could be a separate issue) and any repeats of an event (ie 11111) are failing to sound. In section 3.9, figure 2 of the above linked document, they give a 911 example. Here all but the last event have the E bit set. When I set the bit for all numbers, I never get an event to sound. I have thought of a couple possible thing but do not know if they are the reason: 1) figure 2 shows payload types of 96 and 97 sent. I have not nor know how to exactly. In section 3.8, codes 96 and 97 are described as "the dynamic payload types 96 and 97 have been assigned for the redundancy mechanism and the telephone event payload respectively" 2) In section 3.5, "E:", "A sender MAY delay setting the end bit until retransmitting the last packet for a tone, rather than on its first transmission" Does anyone have an idea of how to actually do this? I have also fiddled around with timestamp intervals and the RTP marker. Any help is greatly appreciated. Here is a sample wireshark event capture of the relevant areas: 6590 31.159045000 xx.x.x.xxx --.--.---.-- RTP EVENT Payload type=RTP Event, DTMF Pound # (end) Real-Time Transport Protocol Stream setup by SDP (frame 6225) Setup frame: 6225 Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 0 Extended sequence number: 65536 Timestamp: 0 Synchronization Source identifier: 0x15f27104 (368210180) RFC 2833 RTP Event Event ID: DTMF Pound # (11) 1... .... = End of Event: True .0.. .... = Reserved: False ..00 0000 = Volume: 0 Event Duration: 2048

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  • Change Call Screen

    - by vyana
    I need to change or customize the call screen when initiating a call on Android. After searching on google I do not find any way to do it. There is no way to send DTMF tones during a call, the idea is to send a specific number to the call screen. So when a call is made is possible to see the number to dial during a call to the PBX. I tried to putting the number in the "status bar", but the notification hide after seconds and it is not practical. I appreciate any other suggestion. Thanks

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  • How do I send DTMF tones and pauses using Android ACTION_CALL Intent with commas in the number?

    - by Rob Kent
    I have an application that calls a number stored by the user. Everything works okay unless the number contains commas or hash signs, in which case the Uri gets truncated after the digits. I have read that you need to encode the hash sign but even doing that, or without a hash sign, the commas never get passed through. However, they do get passed through if you just pick the number from your contacts. I must be doing something wrong. For example: String number = "1234,,,,4#1"; Uri uri = Uri.parse(String.format("tel:%s", number)); try { startActivity(new Intent(callType, uri)); } catch (ActivityNotFoundException e) { ... Only the number '1234' would end up in the dialer.

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  • Skype keypad tones

    - by Don
    Hi, When I push a number on the Skype keypad (or use the number keys on the keyboard) no tone is emitted. This happens both when dialling a number and if I push a key during a call. This makes it impossible for me to use Skype with automated telephone systems that require you to use the keypad to enter data or choose between various options. I spoke to somebody who works in a call centre about this and they indicated that somebody had mentioned that it's possible to disable (DTMF) tones in Skype. I've looked through all the Skype options and can't find any way to enable/disable DTMF tones. If somebody knows how I can do this, or has another suggestion for fixing the problem, please let me know. I'm using version 4.2.0.152 of Skype. Thanks, Don

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  • Show dialog on outbound calls

    - by jgauffin
    I want to be able to show a dialog on outbound calls. The dialog is used to ask the user if he wants 1. dial the phone number directly 2. Dial through the PBX. If option two is chosen, i want to dial a specific number and send the dialed number as DTMF. How do I catch and stop outgoing calls? How do I get the dialed number?

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  • Playing an arbitrary tone with Android.

    - by fiXedd
    Is there any way to make Android emit a sound of arbitrary frequency (meaning, I don't want to have pre-recorded sound files)? I've looked around and ToneGenerator was the only thing I was able to find that was even close, but it seems to only be capable of outputting the standard DTMF tones. Any ideas?

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  • Adding or reading some of the contact fields in sybian using j2me

    - by learn
    I want to add or read the fields of cotact like i am getting the telephone home no ContactList clist; Contact con; String no; if(cList.isSupportedAttribute(Contact.TEL, Contact.ATTR_HOME)) { con.addString(Contact.TEL, Contact.ATTR_HOME, no); } and mobile no if(cList.isSupportedAttribute(Contact.TEL, Contact.ATTR_MOBILE)) { con.addString(Contact.TEL, Contact.ATTR_MOBILE, mb); } now i want to get the fields internet telephone, push to talk, mobile(home), mobile(business), dtmf, shareview, sip, children, spouse and some more fields please help me.. Thanks in advance

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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  • Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

    - by MasterRoot24
    I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN ports connected to FE4 WAN on my Cisco 881. The Cisco 881 get's a DHCP provided IP from my ISP. My LAN is part of default Vlan 1 (192.168.1.0/24). General internet connectivity is working great, I've managed to setup static NAT rules for my HTTP/HTTPS/SMTP/etc. services which are running on my LAN. I don't know whether it's worth mentioning that I've opted to use NVI NAT (ip nat enable as opposed to the traditional ip nat outside/ip nat inside) setup. My reason for this is that NVI allows NAT loopback from my LAN to the WAN IP and back in to the necessary server on the LAN. I run an Asterisk 1.8 PBX on my LAN, which connects to a SIP provider on the internet. Both inbound and outbound calls through the old setup (WAG320N providing routing/NAT) worked fine. However, since moving to the Cisco 881, inbound calls drop after around 10 seconds, whereas outbound calls work fine. The following message is logged on my Asterisk PBX: [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6528ms with no response [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3670 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). (I know that this is quite a common issue - I've spend the best part of 2 days solid on this, trawling Google.) I've done as I am told and checked https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. Referring to the section "Other SIP requests" in the page linked above, I believe that the hangup to be caused by the ACK from my SIP provider not being passed back through NAT to Asterisk on my PBX. I tried to ascertain this by dumping the packets on my WAN interface on the 881. I managed to obtain a PCAP dump of packets in/out of my WAN interface. Here's an example of an ACK being reveived by the router from my provider: 689 21.219999 193.x.x.x 188.x.x.x SIP 502 Request: ACK sip:[email protected] | However a SIP trace on the Asterisk server show's that there are no ACK's received in response to the 200 OK from my PBX: http://pastebin.com/wwHpLPPz In the past, I have been strongly advised to disable any sort of SIP ALGs on routers and/or firewalls and the many posts regarding this issue on the internet seem to support this. However, I believe on Cisco IOS, the config command to disable SIP ALG is no ip nat service sip udp port 5060 however, this doesn't appear to help the situation. To confirm that config setting is set: Router1#show running-config | include sip no ip nat service sip udp port 5060 Another interesting twist: for a short period of time, I tried another provider. Luckily, my trial account with them is still available, so I reverted my Asterisk config back to the revision before I integrated with my current provider. I then dialled in to the DDI associated with the trial trunk and the call didn't get hung up and I didn't get the error above! To me, this points at the provider, however I know, like all providers do, will say "There's no issues with our SIP proxies - it's your firewall." I'm tempted to agree with this, as this issue was not apparent with the old WAG320N router when it was doing the NAT'ing. I'm sure you'll want to see my running-config too: ! ! Last configuration change at 15:55:07 UTC Sun Dec 9 2012 by xxx version 15.2 no service pad service tcp-keepalives-in service tcp-keepalives-out service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone no service password-encryption service sequence-numbers ! hostname Router1 ! boot-start-marker boot-end-marker ! ! security authentication failure rate 10 log security passwords min-length 6 logging buffered 4096 logging console critical enable secret 4 xxx ! aaa new-model ! ! aaa authentication login local_auth local ! ! ! ! ! aaa session-id common ! memory-size iomem 10 ! crypto pki trustpoint TP-self-signed-xxx enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-xxx revocation-check none rsakeypair TP-self-signed-xxx ! ! crypto pki certificate chain TP-self-signed-xxx certificate self-signed 01 quit no ip source-route no ip gratuitous-arps ip auth-proxy max-login-attempts 5 ip admission max-login-attempts 5 ! ! ! ! ! no ip bootp server ip domain name dmz.merlin.local ip domain list dmz.merlin.local ip domain list merlin.local ip name-server x.x.x.x ip inspect audit-trail ip inspect udp idle-time 1800 ip inspect dns-timeout 7 ip inspect tcp idle-time 14400 ip inspect name autosec_inspect ftp timeout 3600 ip inspect name autosec_inspect http timeout 3600 ip inspect name autosec_inspect rcmd timeout 3600 ip inspect name autosec_inspect realaudio timeout 3600 ip inspect name autosec_inspect smtp timeout 3600 ip inspect name autosec_inspect tftp timeout 30 ip inspect name autosec_inspect udp timeout 15 ip inspect name autosec_inspect tcp timeout 3600 ip cef login block-for 3 attempts 3 within 3 no ipv6 cef ! ! multilink bundle-name authenticated license udi pid CISCO881-SEC-K9 sn ! ! username xxx privilege 15 secret 4 xxx username xxx secret 4 xxx ! ! ! ! ! ip ssh time-out 60 ! ! ! ! ! ! ! ! ! interface FastEthernet0 no ip address ! interface FastEthernet1 no ip address ! interface FastEthernet2 no ip address ! interface FastEthernet3 switchport access vlan 2 no ip address ! interface FastEthernet4 ip address dhcp no ip redirects no ip unreachables no ip proxy-arp ip nat enable duplex auto speed auto ! interface Vlan1 ip address 192.168.1.1 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip nat enable ! interface Vlan2 ip address 192.168.0.2 255.255.255.0 ! ip forward-protocol nd ip http server ip http access-class 1 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ! ! no ip nat service sip udp port 5060 ip nat source list 1 interface FastEthernet4 overload ip nat source static tcp x.x.x.x 80 interface FastEthernet4 80 ip nat source static tcp x.x.x.x 443 interface FastEthernet4 443 ip nat source static tcp x.x.x.x 25 interface FastEthernet4 25 ip nat source static tcp x.x.x.x 587 interface FastEthernet4 587 ip nat source static tcp x.x.x.x 143 interface FastEthernet4 143 ip nat source static tcp x.x.x.x 993 interface FastEthernet4 993 ip nat source static tcp x.x.x.x 1723 interface FastEthernet4 1723 ! ! logging trap debugging logging facility local2 access-list 1 permit 192.168.1.0 0.0.0.255 access-list 1 permit 192.168.0.0 0.0.0.255 no cdp run ! ! ! ! control-plane ! ! banner motd Authorized Access only ! line con 0 login authentication local_auth length 0 transport output all line aux 0 exec-timeout 15 0 login authentication local_auth transport output all line vty 0 1 access-class 1 in logging synchronous login authentication local_auth length 0 transport preferred none transport input telnet transport output all line vty 2 4 access-class 1 in login authentication local_auth length 0 transport input ssh transport output all ! ! end ...and, if it's of any use, here's my Asterisk SIP config: [general] context=default ; Default context for calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. directmedia=no ; Don't allow direct RTP media between extensions (doesn't work through NAT) externhost=<MY DYNDNS HOSTNAME> ; Our external hostname to resolve to IP and be used in NAT'ed packets localnet=192.168.1.0/24 ; Define our local network so we know which packets need NAT'ing qualify=yes ; Qualify peers by default dtmfmode=rfc2833 ; Set the default DTMF mode disallow=all ; Disallow all codecs by default allow=ulaw ; Allow G.711 u-law allow=alaw ; Allow G.711 a-law ; ---------------------- ; SIP Trunk Registration ; ---------------------- ; Orbtalk register => <MY SIP PROVIDER USER NAME>:[email protected]/<MY DDI> ; Main Orbtalk number ; ---------- ; Trunks ; ---------- [orbtalk] ; Main Orbtalk trunk type=peer insecure=invite host=sipgw3.orbtalk.co.uk nat=yes username=<MY SIP PROVIDER USER NAME> defaultuser=<MY SIP PROVIDER USER NAME> fromuser=<MY SIP PROVIDER USER NAME> secret=xxx context=inbound I really don't know where to go with this. If anyone can help me find out why these calls are being dropped off, I'd be grateful if you could chime in! Please let me know if any further info is required.

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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