Re: [asterisk-users] Hangup Reason
Hi Alexander Le 06/05/2021 à 17:15, Alexander Perkins a écrit : Hi All. We've put in a check for Do Not Call before a call goes out. However, we have noticed that we cannot seem to pass a 'hangup reason' for a call. For example, I'd like to know that this number is on the DNC so our system does not call them back. Is it possible to pass a hangup reason to Asterisk? Not so much a code, but a reason. Or is there a code for DNC? You should add your own PJSIP headers for that -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
I was able to get on the UI of the Yealink T32G and fiddle with the setting. Here's the setting for TLS transport in /etc/asterisk/extensions.conf: [transport-tls] type = transport protocol = tls bind = 0.0.0.0:5061 ; ca_list_file = /etc/asterisk/keys/ca.crt ; cert_file = /etc/asterisk/keys/asterisk.crt ; priv_key_file = /etc/asterisk/keys/asterisk.key cert_file = /etc/asterisk/keys/fullchain.pem priv_key_file = /etc/asterisk/keys/privkey.pem method = tlsv1_2 allow_reload = true Using FQHN for sip server still results in the same error with the phone failing to registered: [Feb 12 16:55:33] WARNING[2080] pjproject:SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336027900> len: 0 peer: 128.171.77.34:45830 I tried to upload my cert.pem (by Letsencrypt) to the phone as one of the trusted certificates and check "accept only trusted certificates". It didn't help. Nor does unchecking "accept only trusted certificates''. There seem to be some reports in freepbx forum re trouble setting up yearlink phones with tls transport: https://community.freepbx.org/t/tls-freepbx-and-yealink/59174 Yealink's writeup re using security certificates was for certain models/firmware levels, and mine isn't among them. I guess I'll probably have to accept that the few Yealink T32G will not play nice with TLS transport and buy the "sanctioned" models when rolling out the new Asterisk 16.14 server. I may also try my luck with the Cisco 7940/7960 phones that populate most of our offices. Thanks, --Ruisheng On Fri, Feb 12, 2021 at 3:13 PM Ruisheng Peng wrote: > Thanks Joshua for the tip re using hostname rather than IP address when > configuring the phone. It worked nicely on the linphone on my macbookpro > at home. Dialplans are followed faithfully w/o the problems I experienced > earlier. I'll test using the hostname on the Yealink phone next time I'm > in office. > > Thanks, > > --Ruisheng > > On Fri, Feb 12, 2021 at 4:48 AM Joshua C. Colp wrote: > >> On Thu, Feb 11, 2021 at 9:01 PM Ruisheng Peng >> wrote: >> >>> Sorry, my bad. I failed to change the transport to tls on the provision >>> for the hardphone, nor did change the transport on the linphone setup. >>> However, after I do that, the hardphone (Yealink T32G) failed to register, >>> citing: >>> >>> [Feb 11 14:16:03] WARNING[24936]: pjproject: : >>> SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336027900> >> routines-SSL23_GET_CLIENT_HELLO-unknown protocol> len: 0 peer: >>> 128.171.77.34:30401 >>> >> >> This would be caused by the TLS transport configuration on Asterisk or >> the phone potentially. You'd need to provide the transport definition from >> pjsip.conf. Without that I can say the "method" option is likely needing >> changing. I'm not familiar with what is supported by Yealink. >> >> >>> on the linphone side, it also fails to register: >>> >>> 2021-02-11 13:26:32:637 [linphone/belle-sip] MESSAGE Trying to connect >>> to [TLS://:::128.171.77.23:5061] >>> >>> 2021-02-11 13:26:32:652 [linphone/belle-sip] MESSAGE Channel >>> [0x7fc8b800]: Connected at TCP level, now doing TLS handshake with >>> cname=128.171.77.23 >>> >>> 2021-02-11 13:26:32:654 [linphone/belle-sip] MESSAGE Channel >>> [0x7fc8b800]: SSL handshake in progress... >>> >>> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >>> depth=[2], flags=[]: >>> >>> cert. version : 3 >>> >>> serial number : 44:AF:B0:80:D6:A3:27:BA:89:30:39:86:2E:F8:40:6B >>> >>> issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 >>> >>> subject name : O=Digital Signature Trust Co., CN=DST Root CA X3 >>> >>> issued on: 2000-09-30 21:12:19 >>> >>> expires on: 2021-09-30 14:01:15 >>> >>> signed using : RSA with SHA1 >>> >>> RSA key size : 2048 bits >>> >>> basic constraints : CA=true >>> >>> key usage : Key Cert Sign, CRL Sign >>> >>> >>> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >>> depth=[1], flags=[]: >>> >>> cert. version : 3 >>> >>> serial number : 40:01:75:04:83:14:A4:C8:21:8C:84:A9:0C:16:CD:DF >>> >>> issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 >>> >>> subject name : C=US, O=Let's Encrypt, CN=R3 >>> >>> issued on: 2020-10-07 19:21:40 >>> >>> expires on: 2021-09-29 19:21:40 >>> >>> signed using : RSA with SHA-256 >>> >>> RSA key size : 2048 bits >>> >>> basic constraints : CA=true, max_pathlen=0 >>> >>> key usage : Digital Signature, Key Cert Sign, CRL Sign >>> >>> ext key usage : TLS Web Server Authentication, TLS Web Client >>> Authentication >>> >>> >>> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >>> depth=[0], flags=[CN-mismatch ]: >>> >>> cert. version : 3 >>> >>> serial number : 03:F0:83:3C:5D:41:76:BC:4E:B2:E6:AB:60:8C:F9:5E:27:86 >>> >>> issuer name : C=US, O=Let's Encrypt, CN=R3 >>> >>> subject name :
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
Thanks Joshua for the tip re using hostname rather than IP address when configuring the phone. It worked nicely on the linphone on my macbookpro at home. Dialplans are followed faithfully w/o the problems I experienced earlier. I'll test using the hostname on the Yealink phone next time I'm in office. Thanks, --Ruisheng On Fri, Feb 12, 2021 at 4:48 AM Joshua C. Colp wrote: > On Thu, Feb 11, 2021 at 9:01 PM Ruisheng Peng > wrote: > >> Sorry, my bad. I failed to change the transport to tls on the provision >> for the hardphone, nor did change the transport on the linphone setup. >> However, after I do that, the hardphone (Yealink T32G) failed to register, >> citing: >> >> [Feb 11 14:16:03] WARNING[24936]: pjproject: :SSL >> SSL_ERROR_SSL (Handshake): Level: 0 err: <336027900> > routines-SSL23_GET_CLIENT_HELLO-unknown protocol> len: 0 peer: >> 128.171.77.34:30401 >> > > This would be caused by the TLS transport configuration on Asterisk or the > phone potentially. You'd need to provide the transport definition from > pjsip.conf. Without that I can say the "method" option is likely needing > changing. I'm not familiar with what is supported by Yealink. > > >> on the linphone side, it also fails to register: >> >> 2021-02-11 13:26:32:637 [linphone/belle-sip] MESSAGE Trying to connect to >> [TLS://:::128.171.77.23:5061] >> >> 2021-02-11 13:26:32:652 [linphone/belle-sip] MESSAGE Channel >> [0x7fc8b800]: Connected at TCP level, now doing TLS handshake with >> cname=128.171.77.23 >> >> 2021-02-11 13:26:32:654 [linphone/belle-sip] MESSAGE Channel >> [0x7fc8b800]: SSL handshake in progress... >> >> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >> depth=[2], flags=[]: >> >> cert. version : 3 >> >> serial number : 44:AF:B0:80:D6:A3:27:BA:89:30:39:86:2E:F8:40:6B >> >> issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 >> >> subject name : O=Digital Signature Trust Co., CN=DST Root CA X3 >> >> issued on: 2000-09-30 21:12:19 >> >> expires on: 2021-09-30 14:01:15 >> >> signed using : RSA with SHA1 >> >> RSA key size : 2048 bits >> >> basic constraints : CA=true >> >> key usage : Key Cert Sign, CRL Sign >> >> >> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >> depth=[1], flags=[]: >> >> cert. version : 3 >> >> serial number : 40:01:75:04:83:14:A4:C8:21:8C:84:A9:0C:16:CD:DF >> >> issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 >> >> subject name : C=US, O=Let's Encrypt, CN=R3 >> >> issued on: 2020-10-07 19:21:40 >> >> expires on: 2021-09-29 19:21:40 >> >> signed using : RSA with SHA-256 >> >> RSA key size : 2048 bits >> >> basic constraints : CA=true, max_pathlen=0 >> >> key usage : Digital Signature, Key Cert Sign, CRL Sign >> >> ext key usage : TLS Web Server Authentication, TLS Web Client >> Authentication >> >> >> 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate >> depth=[0], flags=[CN-mismatch ]: >> >> cert. version : 3 >> >> serial number : 03:F0:83:3C:5D:41:76:BC:4E:B2:E6:AB:60:8C:F9:5E:27:86 >> >> issuer name : C=US, O=Let's Encrypt, CN=R3 >> >> subject name : CN=voip1.ifa.hawaii.edu >> >> issued on: 2020-12-30 02:56:29 >> >> expires on: 2021-03-30 02:56:29 >> >> signed using : RSA with SHA-256 >> >> RSA key size : 2048 bits >> >> basic constraints : CA=false >> >> subject alt name : voip1.ifa.hawaii.edu >> >> key usage : Digital Signature, Key Encipherment >> >> ext key usage : TLS Web Server Authentication, TLS Web Client >> Authentication >> >> >> 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Channel >> [0x7fc8b800]: SSL handshake failed : X509 - Certificate verification >> failed, e.g. CRL, CA or signature check failed >> >> 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Cannot connect to >> [TLS://128.171.77.23:5061] >> > > I don't use linphone or have any experience so can only provide general > comments. Either the certificate chain is incomplete and the client can't > verify, or the client doesn't have the certificate authority root > certificate as trusted. As well if you aren't doing so you have to connect > to the hostname - you can't specify the IP address. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users --
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
On Thu, Feb 11, 2021 at 9:01 PM Ruisheng Peng wrote: > Sorry, my bad. I failed to change the transport to tls on the provision > for the hardphone, nor did change the transport on the linphone setup. > However, after I do that, the hardphone (Yealink T32G) failed to register, > citing: > > [Feb 11 14:16:03] WARNING[24936]: pjproject: :SSL > SSL_ERROR_SSL (Handshake): Level: 0 err: <336027900> routines-SSL23_GET_CLIENT_HELLO-unknown protocol> len: 0 peer: > 128.171.77.34:30401 > This would be caused by the TLS transport configuration on Asterisk or the phone potentially. You'd need to provide the transport definition from pjsip.conf. Without that I can say the "method" option is likely needing changing. I'm not familiar with what is supported by Yealink. > on the linphone side, it also fails to register: > > 2021-02-11 13:26:32:637 [linphone/belle-sip] MESSAGE Trying to connect to > [TLS://:::128.171.77.23:5061] > > 2021-02-11 13:26:32:652 [linphone/belle-sip] MESSAGE Channel > [0x7fc8b800]: Connected at TCP level, now doing TLS handshake with > cname=128.171.77.23 > > 2021-02-11 13:26:32:654 [linphone/belle-sip] MESSAGE Channel > [0x7fc8b800]: SSL handshake in progress... > > 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate > depth=[2], flags=[]: > > cert. version : 3 > > serial number : 44:AF:B0:80:D6:A3:27:BA:89:30:39:86:2E:F8:40:6B > > issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 > > subject name : O=Digital Signature Trust Co., CN=DST Root CA X3 > > issued on: 2000-09-30 21:12:19 > > expires on: 2021-09-30 14:01:15 > > signed using : RSA with SHA1 > > RSA key size : 2048 bits > > basic constraints : CA=true > > key usage : Key Cert Sign, CRL Sign > > > 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate > depth=[1], flags=[]: > > cert. version : 3 > > serial number : 40:01:75:04:83:14:A4:C8:21:8C:84:A9:0C:16:CD:DF > > issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 > > subject name : C=US, O=Let's Encrypt, CN=R3 > > issued on: 2020-10-07 19:21:40 > > expires on: 2021-09-29 19:21:40 > > signed using : RSA with SHA-256 > > RSA key size : 2048 bits > > basic constraints : CA=true, max_pathlen=0 > > key usage : Digital Signature, Key Cert Sign, CRL Sign > > ext key usage : TLS Web Server Authentication, TLS Web Client > Authentication > > > 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate > depth=[0], flags=[CN-mismatch ]: > > cert. version : 3 > > serial number : 03:F0:83:3C:5D:41:76:BC:4E:B2:E6:AB:60:8C:F9:5E:27:86 > > issuer name : C=US, O=Let's Encrypt, CN=R3 > > subject name : CN=voip1.ifa.hawaii.edu > > issued on: 2020-12-30 02:56:29 > > expires on: 2021-03-30 02:56:29 > > signed using : RSA with SHA-256 > > RSA key size : 2048 bits > > basic constraints : CA=false > > subject alt name : voip1.ifa.hawaii.edu > > key usage : Digital Signature, Key Encipherment > > ext key usage : TLS Web Server Authentication, TLS Web Client > Authentication > > > 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Channel > [0x7fc8b800]: SSL handshake failed : X509 - Certificate verification > failed, e.g. CRL, CA or signature check failed > > 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Cannot connect to > [TLS://128.171.77.23:5061] > I don't use linphone or have any experience so can only provide general comments. Either the certificate chain is incomplete and the client can't verify, or the client doesn't have the certificate authority root certificate as trusted. As well if you aren't doing so you have to connect to the hostname - you can't specify the IP address. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
Sorry, my bad. I failed to change the transport to tls on the provision for the hardphone, nor did change the transport on the linphone setup. However, after I do that, the hardphone (Yealink T32G) failed to register, citing: [Feb 11 14:16:03] WARNING[24936]: pjproject: :SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336027900> len: 0 peer: 128.171.77.34:30401 on the linphone side, it also fails to register: 2021-02-11 13:26:32:637 [linphone/belle-sip] MESSAGE Trying to connect to [TLS://:::128.171.77.23:5061] 2021-02-11 13:26:32:652 [linphone/belle-sip] MESSAGE Channel [0x7fc8b800]: Connected at TCP level, now doing TLS handshake with cname=128.171.77.23 2021-02-11 13:26:32:654 [linphone/belle-sip] MESSAGE Channel [0x7fc8b800]: SSL handshake in progress... 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate depth=[2], flags=[]: cert. version : 3 serial number : 44:AF:B0:80:D6:A3:27:BA:89:30:39:86:2E:F8:40:6B issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 subject name : O=Digital Signature Trust Co., CN=DST Root CA X3 issued on: 2000-09-30 21:12:19 expires on: 2021-09-30 14:01:15 signed using : RSA with SHA1 RSA key size : 2048 bits basic constraints : CA=true key usage : Key Cert Sign, CRL Sign 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate depth=[1], flags=[]: cert. version : 3 serial number : 40:01:75:04:83:14:A4:C8:21:8C:84:A9:0C:16:CD:DF issuer name : O=Digital Signature Trust Co., CN=DST Root CA X3 subject name : C=US, O=Let's Encrypt, CN=R3 issued on: 2020-10-07 19:21:40 expires on: 2021-09-29 19:21:40 signed using : RSA with SHA-256 RSA key size : 2048 bits basic constraints : CA=true, max_pathlen=0 key usage : Digital Signature, Key Cert Sign, CRL Sign ext key usage : TLS Web Server Authentication, TLS Web Client Authentication 2021-02-11 13:26:32:674 [linphone/belle-sip] MESSAGE Found certificate depth=[0], flags=[CN-mismatch ]: cert. version : 3 serial number : 03:F0:83:3C:5D:41:76:BC:4E:B2:E6:AB:60:8C:F9:5E:27:86 issuer name : C=US, O=Let's Encrypt, CN=R3 subject name : CN=voip1.ifa.hawaii.edu issued on: 2020-12-30 02:56:29 expires on: 2021-03-30 02:56:29 signed using : RSA with SHA-256 RSA key size : 2048 bits basic constraints : CA=false subject alt name : voip1.ifa.hawaii.edu key usage : Digital Signature, Key Encipherment ext key usage : TLS Web Server Authentication, TLS Web Client Authentication 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Channel [0x7fc8b800]: SSL handshake failed : X509 - Certificate verification failed, e.g. CRL, CA or signature check failed 2021-02-11 13:26:32:674 [linphone/belle-sip] ERROR Cannot connect to [TLS:// 128.171.77.23:5061] On Mon, Feb 8, 2021 at 12:27 PM Joshua C. Colp wrote: > On Mon, Feb 8, 2021 at 6:14 PM Ruisheng Peng wrote: > >> Thanks Jashua for the suggestion. To find out if the issue was only >> limited to the softphone that was using tls transport (SOFTPHONE_B on ext >> 103, a linphone running off my MBP), I also turned one of the hard phone >> (f30A0A01 on ext 100, a Yealink T32G) into using tls transport. It >> behaves similarly to the linphone in that the Hangup() call in dialplan is >> silently ignored, and the handsets would alway appear as busy/unavilable. >> > > Have you configured the devices, on them or using their provisioning, to > use TLS? It does not appear so as they are using UDP, while you're forcing > a TLS transport in Asterisk. This would not work. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
On Mon, Feb 8, 2021 at 6:14 PM Ruisheng Peng wrote: > Thanks Jashua for the suggestion. To find out if the issue was only > limited to the softphone that was using tls transport (SOFTPHONE_B on ext > 103, a linphone running off my MBP), I also turned one of the hard phone > (f30A0A01 on ext 100, a Yealink T32G) into using tls transport. It > behaves similarly to the linphone in that the Hangup() call in dialplan is > silently ignored, and the handsets would alway appear as busy/unavilable. > Have you configured the devices, on them or using their provisioning, to use TLS? It does not appear so as they are using UDP, while you're forcing a TLS transport in Asterisk. This would not work. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
Thanks Jashua for the suggestion. To find out if the issue was only
limited to the softphone that was using tls transport (SOFTPHONE_B on ext
103, a linphone running off my MBP), I also turned one of the hard phone
(f30A0A01 on ext 100, a Yealink T32G) into using tls transport. It
behaves similarly to the linphone in that the Hangup() call in dialplan is
silently ignored, and the handsets would alway appear as busy/unavilable.
Here're the relevant part of my /etc/asterisk/extensions.conf:
[globals]
; General internal dialing options used in context Dial-Users.
; Only the timeout is defined here. See the Dial app documentation for
; additional options.
INTERNAL_DIAL_OPT=,30
RP_Yealink = PJSIP/f30A0A01
RP_Cisco = PJSIP/f30B0B02
RP_HMBP = PJSIP/SOFTPHONE_A
RP_OMBP = PJSIP/SOFTPHONE_B
[sets]
exten => 100,1,Dial(${RP_Yealink},10,m)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 101,1,Dial(${RP_Cisco},10)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 102,1,Dial(${RP_HMBP})
exten => 103,1,Dial(${RP_OMBP},10)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 110,1,Dial(${RP_Yealink}&${RP_Cisco})
exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
Here're what pjsip logger captures when using the tls softphone (on ext
103) to call ext 101 (Hello World!). I had to click the hanup button on the
linphone some 15s later to terminate the call.
<--- Received SIP request (1199 bytes) from UDP:128.171.168.233:5060 --->
INVITE sip:200@128.171.77.23 SIP/2.0
Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.D-YbrxKYs;rport
From: "VOIP1_test" ;tag=XvCbVpnIJ
To: sip:200@128.171.77.23
CSeq: 20 INVITE
Call-ID: ziUzVUxYw7
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 531
Contact: ;expires=3599;+sip.instance=""
User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88
v=0
o=SOFTPHONE_B 1261 3707 IN IP4 128.171.168.233
s=Talk
c=IN IP4 128.171.168.233
t=0 0
a=rtcp-xr:rcvr-rtt=all:1 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
<--- Transmitting SIP response (479 bytes) to UDP:128.171.168.233:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 128.171.168.233:5060
;rport=5060;received=128.171.168.233;branch=z9hG4bK.D-YbrxKYs
Call-ID: ziUzVUxYw7
From: "VOIP1_test" ;tag=XvCbVpnIJ
To: ;tag=z9hG4bK.D-YbrxKYs
CSeq: 20 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7",opaque="50221ed627077186",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.14.0
Content-Length: 0
<--- Received SIP request (412 bytes) from UDP:128.171.168.233:5060 --->
ACK sip:200@128.171.77.23 SIP/2.0
Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.D-YbrxKYs;rport
Call-ID: ziUzVUxYw7
From: "VOIP1_test" ;tag=XvCbVpnIJ
To: ;tag=z9hG4bK.D-YbrxKYs
Contact: ;expires=3599;+sip.instance=""
Max-Forwards: 70
CSeq: 20 ACK
<--- Received SIP request (1484 bytes) from UDP:128.171.168.233:5060 --->
INVITE sip:200@128.171.77.23 SIP/2.0
Via: SIP/2.0/UDP 128.171.168.233:5060;branch=z9hG4bK.HgO8RDlH4;rport
From: "VOIP1_test" ;tag=XvCbVpnIJ
To: sip:200@128.171.77.23
CSeq: 21 INVITE
Call-ID: ziUzVUxYw7
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 531
Contact: ;expires=3599;+sip.instance=""
User-Agent: Linphone Desktop/4.2.2 (macOS 10.15, Qt 5.14.2)
LinphoneCore/4.4.0-13-gc99cb9c88
Authorization: Digest realm="asterisk",
nonce="1612573994/b1f976725d3cbb6b1fc9af5923a87ac7", algorithm=md5,
opaque="50221ed627077186", username="SOFTPHONE_B", uri="
sip:200@128.171.77.23", response="352ca45cd5adc103f4b679713905bde9",
cnonce="7F142IC~o5UVxMll", nc=0001, qop=auth
v=0
o=SOFTPHONE_B 1261 3707 IN IP4 128.171.168.233
s=Talk
c=IN IP4 128.171.168.233
t=0 0
a=rtcp-xr:rcvr-rtt=all:1 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
== Setting global variable 'SIPDOMAIN' to
Re: [asterisk-users] Hangup() not working for handsets using pls transport?
On Wed, Feb 3, 2021 at 11:02 PM Ruisheng Peng wrote:
When using handsets with udp or tcp transports to dial ext 100, it'd hangup
> after the no-one-arround message. However, when using the handset with tls
> transport, it doesn't hang up on its own if ext 100 is not answered. I
> have to click the hangup button to accomplish that. Here's what asterisk
> log shows:
>
> == Setting global variable 'SIPDOMAIN' to '128.171.77.23'
>
> -- Executing [100@sets:1] Dial("PJSIP/SOFTPHONE_B-0007", "
> PJSIP/f30A0A01,10,m") in new stack
>
> -- Called PJSIP/f30A0A01
>
> -- Started music on hold, class 'default', on channel
> 'PJSIP/SOFTPHONE_B-0007'
>
>> 0x7f0fa801ede0 -- Strict RTP learning after remote address set
> to: 128.171.168.233:7078
>
> -- PJSIP/f30A0A01-0008 is ringing
>
> -- PJSIP/f30A0A01-0008 is ringing
>
>> 0x7f0fa801ede0 -- Strict RTP switching to RTP target address
> 128.171.168.233:7078 as source
>
>> 0x7f0fa801ede0 -- Strict RTP learning complete - Locking on
> source address 128.171.168.233:7078
>
> -- Nobody picked up in 1 ms
>
> -- Stopped music on hold on PJSIP/SOFTPHONE_B-0007
>
> -- Executing [100@sets:2] Playback("PJSIP/SOFTPHONE_B-0007", "
> vm-nobodyavail") in new stack
>
> -- Playing 'vm-nobodyavail.slin'
> (language 'en')
>
> -- Executing [100@sets:3] Hangup("PJSIP/SOFTPHONE_B-0007", "") in
> new stack
>
> == Spawn extension (sets, 100, 3) exited non-zero on
> 'PJSIP/SOFTPHONE_B-0007'
> voip1*CLI>
>
> Another quirk is when I use a phone with udp transport (RP_Yealink) to
> call a phone with tls transport (RP_OMBP) it immediately jumps
> the no-one-around message w/o ringing, then hang up. The tls phone is
> shown available but asterisk sees it busy:
>
> == Setting global variable 'SIPDOMAIN' to '128.171.77.23'
>
> -- Executing [103@sets:1] Dial("PJSIP/f30A0A01-000d", "
> PJSIP/SOFTPHONE_B,10") in new stack
>
> -- Called PJSIP/SOFTPHONE_B
>
> == Everyone is busy/congested at this time (1:0/1/0)
>
> -- Executing [103@sets:2] Playback("PJSIP/f30A0A01-000d", "
> vm-nobodyavail") in new stack
>
>> 0x7f0fa000c330 -- Strict RTP learning after remote address set
> to: 128.171.77.118:11790
>
>> 0x7f0fa000c330 -- Strict RTP switching to RTP target address
> 128.171.77.118:11790 as source
>
> -- Playing 'vm-nobodyavail.slin'
> (language 'en')
>
> -- Executing [103@sets:3] Hangup("PJSIP/f30A0A01-000d", "")
> in new stack
>
> == Spawn extension (sets, 103, 3) exited non-zero on
> 'PJSIP/f30A0A01-000d'
>
> voip1*CLI>
>
> Suppose it's not cool to mix transports among your handsets? Any
> suggestions?
>
I'd suggest looking at the actual SIP signaling to see what is going on
using "pjsip set logger on" and also providing configuration. This would
allow better insight into what exactly is going on.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Hangup-handler on failed calls
Found a workaround... In case anyone else runs into something similar:
Setting congestion=yes in cdr.conf changes the writing behavior, and
instead of having one CDR with disposition=FAILED, I have all the CDRs with
disposition=CONGESTION, and as I can link them together with the
linkedid or the uniqueid plus the sequence, I can grab the fields from the
row that has them.
On Tue, Feb 25, 2020 at 4:01 PM Joel Serrano wrote:
> Hello,
>
> I have a setup with asterisk 16.8.0, I'm facing a problem where calls that
> fail (CONGESTION) don't have filled in some extra fields we add to the CDRs
> in the database.
>
> We use cdr_adaptive_odbc with MySQL as backend.
>
> To simplify the scenario:
>
> [sub-hanguphandler]
> exten => s,1,Set(CDR(foo)=${bar})
> same => n,Return()
>
> [default]
> exten => _X.,1,NoOp(New test call)
> same => n,Set(CHANNEL(hangup_handler_push)=sub-hanguphandler,s,1)
> same => n,Dial(...)
> same => n,Hangup()
>
>
> With the above config:
>
> 1- An answered call: foo is filled in db
> 2- A cancelled call: foo is filled in db
> 3- A failed call: foo is NOT filled in db
>
> In all 3 cases, with verbose logs enabled I can see
> the sub-hanguphandler subroutine is being executed, so it's confusing.
>
> I would understand if the hangup handler is NOT executed for failed calls,
> but seeing it in the logs and then seeing the field empty in the db doesn't
> make sense to me.
>
> Any tips on where/how I can troubleshoot this?
>
>
> Thanks,
> Joel.
>
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Re: [asterisk-users] Hangup hook to put back a call into a queue
It might work for you to branch on ${DIALSTRING} just after your Dial
command, if you want to handle a BUSY, NOANSWER, or other result. But if
the peer of that Dial hungup, then based on what Joshua said, it seems
there's no recovery.
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Re: [asterisk-users] Hangup hook to put back a call into a queue
On Wed, Feb 5, 2020 at 12:34 PM Farkas Levente wrote:
> hi,
> I hope someone can help me:-)
> we’ve got a freepbx server. there are 2 special extensions (2001, 2002).
> if someone calls this extensions (or a call is forwarded to these
> extensions) and these extension hangup (not the caller party), then we’d
> like to put the calls back into a queue (1000) and wouldn’t like to hangup.
>
> I read your description about hangup hooks:
> https://community.freepbx.org/t/hooking-for-fun-and-income/57718
>
> but still not able to implement it:-(
> what I’ve done:
> * found out in a hard way how to detect the current destination
> extension (because it’s turn out that CALLERID(dnid) is not working in
> case of forwarded call it’s show the original destination)
> * write a macro-dialout-one-predial-hook and a hook marco like this:
>
> [macro-dialout-one-predial-hook]
> exten => s,1,Noop(Entering user defined context
> macro-dialout-one-predial-hook in extensions_custom.conf)
> exten => s,n,GotoIf($["${DEXTEN}"=“2001”]?special)
> exten => s,n,GotoIf($["${DEXTEN}"=“2002”]?special)
> exten => s,n,MacroExit
> exten => s,n(special),NoOp(--- Push Special Hangup Handler
> --)
> exten => s,n,Set(CHANNEL(hangup_handler_push)=back-to-1000-hangup,s,1)
> exten => s,n,MacroExit
>
> [back-to-1000-hangup]
> exten => s,1,Noop(== Entering user defined context
> back-to-1000-hangup ===)
> exten => s,n,Queue(1000)
> exten => s,n,Return
>
> it seems to be called and seem to enter into to call but immediately
> hangup.
> first of all, in this case when in the hangup handler I will NOT like to
> hangup how should I finish the marco?:
>
Hangup handlers don't allow you any control over the hangup process. You
can't stop it from occurring and in fact when it occurs the channel is
already hung up. Anything that expects a live channel won't work.
> Hangup
> Return
> MacroExit
> how to redirect the call to the queue?:
>
> Queue(1000)
> ChannelRedirect(${CHANNEL},,1000,1)
> Gosub(ext-intercom,*801000,1())
> dial-one,HhTtrM(auto-blkvm),1000
> and what is the reason I can’t put the call back to the queue?
> I know that I'm already in the hangup sequence, but still wouldn't like
> to hangup.
> or this can't be done in the hangup handler?
>
You can't do it from a hangup handler. The Dial option provides the g
option[1] which can be used to continue dialplan execution when the called
party hangs up, but I don't work on FreePBX so I can't comment on how best
to use it there.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial
--
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Asterisk Technical Lead
Sangoma Technologies
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Re: [asterisk-users] Hangup handler gosub error with asterisk 16.4.0.
On 6/1/19 9:18 AM, Harley Peters wrote:
I am receiving the following errors on any hangup handler subroutines.
[2019-05-31 18:22:13.958] VERBOSE[23943][C-0009] app_stack.c:
PJSIP/104090401-000a Internal Gosub(PreventForwardingLoop,s,1)) start
[2019-05-31 18:22:13.958] NOTICE[23943][C-0009] pbx.c: No such label
'1)' in extension 's' in context 'PreventForwardingLoop'
[2019-05-31 18:22:13.958] WARNING[23943][C-0009] pbx.c: Priority
'1)' must be a number > 0, or valid label
[2019-05-31 18:22:13.958] ERROR[23943][C-0009] app_stack.c: Gosub
address is invalid: 'PreventForwardingLoop,s,1)'
Dialplan:
exten =>
_1NXXNXX,n,Set(CHANNEL(hangup_handler_push)=PreventForwardingLoop,s,1))
[PreventForwardingLoop]
exten =>
s,1,Set(DELETEKEY=${DB_DELETE(PreventForwardingLoop/${USER}/${CALLERID(num)})})
exten => s,n,Return()
This is one example it fails on all of them.
I have no problems with asterisk-16.2.1 and earlier.
Any idea what the problem is or is this a bug?
Harley Peters
Well now I just feel stupid.
There's an extra closing parenthesis
exten =>
_1NXXNXX,n,Set(CHANNEL(hangup_handler_push)=PreventForwardingLoop,s,1))
<- shouldn't be there.
It must have been running okay this way for years.
Harley Peters
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Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:32 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file.
Define "send". How are you doing it?
GoSub(play-long-file,s,1)
You can't have a channel both in dialplan directly and also bridged to another
channel at the same time. There's not enough context or information to really
be able to answer without understanding fully.
Maybe this will help explain it. Here's the cli:
Executing [s@incoming:7] Dial("SIP/incall-0001",
"DAHDI/g0,55,tTD(:1)") in new stack
-- Called DAHDI/g0
-- DAHDI/1-1 answered SIP/incall-0001
-- Channel DAHDI/1-1 joined 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel SIP/incall-0001 joined 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- SIP/incall-0001 Internal Gosub(long-file,s,1) start
-- Executing [s@long-file:1] Playback("SIP/incall-0001",
"long-file") in new stack
-- Playing 'long-file.slin' (language
'en')
-- Executing [s@long-file:2] Verbose("SIP/incall-0001",
"bridgepeer is DAHDI/1-1") in new stack
Executing [s@long-file:3] Hangup("SIP/incall-0001", "") in new stack
== Spawn extension (long-file, s, 3) exited non-zero on
'SIP/incall-0001'
[Sep 12 13:06:06] NOTICE[2217][C-0001]: app_stack.c:1082 gosub_run:
SIP/callcentric20-0001 Abnormal 'Gosub(long-file,s,1)' exit.
Popping routine return locations.
-- Channel SIP/incall left 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel DAHDI/1-1 left 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
As you can see DAHDI/1-1 is not hungup until after Playback. I want to
hangup DAHDI/1-1 before the Playback.
Thanks,
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Re: [asterisk-users] hangup the _called_ channel ?
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote: > On 9/12/18 1:22 PM, Joshua Colp wrote: > > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: > >> I understand that HangUp() hangs up the calling channel. I want to > >> hangup the called channel. > >> > >> SIP/mycall-x calls and bridges with DAHDI/1-1. > >> > >> I send SIP/ to listen to a long, very long, file. > > > > Define "send". How are you doing it? > > > GoSub(play-long-file,s,1) You can't have a channel both in dialplan directly and also bridged to another channel at the same time. There's not enough context or information to really be able to answer without understanding fully. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:25 PM, sean darcy wrote: On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? And I'm using dynamic features, applicationmap. play-file=*8,peer,GoSub,"pay-long-file,s,1" -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: > I understand that HangUp() hangs up the calling channel. I want to > hangup the called channel. > > SIP/mycall-x calls and bridges with DAHDI/1-1. > > I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup call gw FXO
2015-03-05 6:11 GMT-06:00 Steve Davies davies...@gmail.com: Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs I tried to contact support sangoma, I send a log to them and they have not contacted me! ,a disappointment asterisk shows active channels, zombie type ;) , for example the extension 160 call the 122, 122 is not connected and tells me this on the phone , I have the impression that rtptimeout not working as it should http://pastebin.com/vTZ0WGqq look cli asterisk: 200.62.89.140(None) koV6foZnHTr3gEf (nothing) No Rx: REGISTER guest 200.62.89.140(None) 690e01185aa2f36 (nothing)No Rx: REGISTER guest 200.62.89.140gatewayVEGA0010-0C09-6C8EF (ulaw) No Rx: ACKgatewayVEGA 200.62.89.140(None) 5db8c434570dfb9 (nothing)No Rx: REGISTER guest 190.184.84.10(None) 3654c4f8-1fd27d (nothing)No Rx: NOTIFY guest 5 active SIP dialogs regardss -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup call gw FXO
On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote: I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking that might help rpt timeout , I have put in 30s, but does not work any advice? regardss something strange, I have some extensions not connected to Asterisk and if I call, I get the message busy, the version I'm using is asterisk 11.15 -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup call gw FXO
Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez xserverli...@gmail.com wrote: On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote: I'm having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Ok my sangoma is a vega 50 , 4 FXO . I tried different tone of countries and does not work, this is the trace of which is for hanging up the channel: http://pastebin.com/y410Rhzt I was thinking that might help rpt timeout , I have put in 30s, but does not work any advice? regardss something strange, I have some extensions not connected to Asterisk and if I call, I get the message busy, the version I'm using is asterisk 11.15 -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Chanel when a peer unregisters
On 04/11/14 15:11, Pat Collins wrote: Hello group and thank you for the attention. I'm using Asterisk 11.12 running on Ubuntu Server 12.04 We have an issue with channels remaining open after a SIP peer unregisters. It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI. Is there any way to drop a channel when the peer using it disappears? Googled every phrase I could think of. No luck. Thank you! Pat Collins rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Chanel when a peer unregisters
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Wednesday, November 05, 2014 4:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hangup Chanel when a peer unregisters On 04/11/14 15:11, Pat Collins wrote: Hello group and thank you for the attention. I'm using Asterisk 11.12 running on Ubuntu Server 12.04 We have an issue with channels remaining open after a SIP peer unregisters. It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI. Is there any way to drop a channel when the peer using it disappears? Googled every phrase I could think of. No luck. Thank you! Pat Collins rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time. Thanks for the response Gareth. The problem is that I may have a conference call up for days at a time. During this time, there may be no activity for hours. If the endpoint the endpoint is able to send RTP keepalive packets, your solution is spot on. Will have a look at it. Thanks again! PC... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup cause 111 after call pickup
On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-1454, and the call is answered. After 7 seconds, the conversation is terminated. [Jun 6 10:13:15] VERBOSE[21118] pbx.c: [Jun 6 10:13:15] -- Executing [120@sub-pickup:25] Pickup(SIP/sipacc3-147c, SIP/SipAgenT01-1454@PICKUPMARK) in new stack [Jun 6 10:13:15] VERBOSE[20788] app_queue.c: [Jun 6 10:13:15] -- SIP/sipacc3-147c answered SIP/SipAgenT01-1454 [Jun 6 10:13:22] VERBOSE[20788] pbx.c: [Jun 6 10:13:22] -- Executing [h@pbx-routing:3] NoOp(SIP/SipAgenT01-1454, hangup cause = 111) in new stack Questions : 1. what can cause a hangup cause 111 ? What is the meaning of hangup cause 111 ? 2. on voip-info.org I read 111 protocol error 500 Server internal error. How can I fix this ?? Using Asterisk 1.8.12.2 on CentOS. Hi Jonas, when the calls is answered, do you have correct both-way audio as well? Please enter sip set debug on on the Asterisk console and paste the output. It could also be helpful if you could paste your dialplan. -- marie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
Hello In indications.com are the tones for several countries On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote: Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
On Tuesday 18 September 2012, Mehdi Rahimi wrote: Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi Unfortunately I am not familiar with the Iranian telephone system. You might have to search for relevant technical standards documentation. For a start, try setting your location to UK -- and if it behaves a bit better, that will be your problem. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup not detected?
Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks like this is a dev issue - I'll start a new thread on the dev mailing list. Justin Killen Senior Programmer / Analyst All American Asphalt 951-736-7600 x 2060 jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, May 24, 2012 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Looks like Swift() (whatever that is) is not returning ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup not detected?
Here is the output from the cli: dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 Since the 'lastapp' variable is 'Swift', this would indicate that the cepstral wrapper is having a problem, correct? Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Tuesday, May 22, 2012 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Okay, the next time it gets in this state I'll gather that information. Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, May 21, 2012 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a time of 16 hours. I'm not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn't seem to progress through the dialplan, they always display the TTS line. Doing a 'dahdi destroy channel 1-1' doesn't seem to be effective - the only way I've been able to clear the calls is to do a 'dahdi restart' and/or restart the asterisk service. For TTS I'm using cepstral with the Swift wrapper. Here is a snippet of my dialplan: Can you post the CLI output of a call that gets hung? I'd like to see where it's hanging on. Also, as a work-around to attempt to solve the symptom and not the underlying issue, you could maybe setup a cron job that runs once every ten minutes that checks for stale calls using AMI, and then hangs up any calls up that are over 10 minutes long? Using the AMI Hangup command? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.comhttp://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] hangup not detected?
Looks like Swift() (whatever that is) is not returning ? On 24 May 2012 23:07, Justin Killen jkil...@allamericanasphalt.com wrote: ** ** ** Here is the output from the cli: ** ** dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect** **Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS ** ** CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 ** ** ** ** ** ** ** ** ** ** Since the ‘lastapp’ variable is ‘Swift’, this would indicate that the cepstral wrapper is having a problem, correct? ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Justin Killen *Sent:* Tuesday, May 22, 2012 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** Okay, the next time it gets in this state I’ll gather that information.*** * ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Monday, May 21, 2012 1:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via ‘core show channel dahdi/1-1’ that have a time of 16 hours. I’m not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn’t seem to progress through the dialplan, they always display the TTS line. Doing a ‘dahdi destroy channel 1-1’ doesn’t seem to be effective – the only way I’ve been able to clear the calls is to do a ‘dahdi restart’ and/or restart the asterisk service.* *** For TTS I’m using cepstral with the Swift wrapper. Here is a snippet of my
Re: [asterisk-users] hangup problem on T1 span
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote: Hello all, I'm trying to solve a problem on a T1 span setup wherein calls are apparently not hanging up properly. CAS or PRI? The system in question is using a Xorcom Astribank with 1 full and 1 partial T1 span, and running Asterisk 1.4.36. The symptom is that when a call hangs up on a DAHDI channel (according to Asterisk), and another outgoing call tries to open a new channel on the same line as the hung-up call within approximately a minute of the hangup, the new call gets a congestion notice (all circuits busy) from asterisk. After about a minute passes after the hangup, the line becomes available again. So it seems like the channels are not hanging up when Asterisk tells them to, and Asterisk doesn't know it. I suspected a signaling issue, and this appeared confirmed when I discovered that the signalling was set in chan_dahdi.conf as fxs_ks (this installation had been converted from analog lines by another company; I guess that was an oversight?). The signalling and such is probably set in /etc/asterisk/dahdi-channels.conf so that setting does not matter. So I changed it to pri_cpe, as my reading of the docs indicated was proper. After this change and restarting everything, though, the symptoms persist. So I figure that either my reading of the docs is wrong (and therefore pri_cpe is not the right signaling) OR something totally unrelated is going on. Can someone please clue me in here? I am a bit at a loss. Let me know if you need further information about the system/environment. What is the output of 'dahdi show channel N' for one such a bad channel when not in a call? Are you sure it's not in a call? See the output of 'core show channels'. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup problem on T1 span
Tzafrir, Thanks for your response. I'll check into those items. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Thu, May 3, 2012 at 4:39 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote: Hello all, I'm trying to solve a problem on a T1 span setup wherein calls are apparently not hanging up properly. CAS or PRI? The system in question is using a Xorcom Astribank with 1 full and 1 partial T1 span, and running Asterisk 1.4.36. The symptom is that when a call hangs up on a DAHDI channel (according to Asterisk), and another outgoing call tries to open a new channel on the same line as the hung-up call within approximately a minute of the hangup, the new call gets a congestion notice (all circuits busy) from asterisk. After about a minute passes after the hangup, the line becomes available again. So it seems like the channels are not hanging up when Asterisk tells them to, and Asterisk doesn't know it. I suspected a signaling issue, and this appeared confirmed when I discovered that the signalling was set in chan_dahdi.conf as fxs_ks (this installation had been converted from analog lines by another company; I guess that was an oversight?). The signalling and such is probably set in /etc/asterisk/dahdi-channels.conf so that setting does not matter. So I changed it to pri_cpe, as my reading of the docs indicated was proper. After this change and restarting everything, though, the symptoms persist. So I figure that either my reading of the docs is wrong (and therefore pri_cpe is not the right signaling) OR something totally unrelated is going on. Can someone please clue me in here? I am a bit at a loss. Let me know if you need further information about the system/environment. What is the output of 'dahdi show channel N' for one such a bad channel when not in a call? Are you sure it's not in a call? See the output of 'core show channels'. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 05:29 PM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code On 04/25/2012 04:45 PM, brya...@zktech.com wrote: Kevin I am using 1.8.x 10.x Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns. Does anyone know what kind of performance hit you take from SIP_CAUSE when you are using few or no calls using chan_local? The performance impact will be directly related to the number of outbound SIP channels you create; no other channels will be involved. We had a Digium OEM customer observe a 50% call load capability decrease when they started using SIP_CAUSE, but that was on a pretty busy system, and all the channels were SIP channels. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has
been reimplemented for Asterisk 11 in a way that doesn't affect
performance as much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Hangup Cause and SIP Response Code
Kevin
I am using 1.8.x 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has been
reimplemented for Asterisk 11 in a way that doesn't affect performance as
much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it
because it is off by default due to performance concerns.
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Flemingkpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what version
of Asterisk you are using. In some versions there is a SIP_CAUSE feature that
can be used to extract that information (although this has been reimplemented
for Asterisk 11 in a way that doesn't affect performance as much as the old
method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Hangup Cause and SIP Response Code
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code On 04/25/2012 04:45 PM, brya...@zktech.com wrote: Kevin I am using 1.8.x 10.x Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns. Does anyone know what kind of performance hit you take from SIP_CAUSE when you are using few or no calls using chan_local? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup after n seconds using originate ?
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like that when creating the originate command? I don't know if it works, but it is worth a shot. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400 Ryan Bullock rrb3...@gmail.com wrote: Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like that when creating the originate command? I don't know if it works, but it is worth a shot. Hi Ryan, thanks for your comment. Unfortunately the 'Variable' parameter is used to push data between the originating script and the dialplan, not commands. Example: Variable: var1=23|var2=24|var3=25 Additionally, this data can be used in the dialplan only when the call gets answered or when it fails. I can't find a way to inject the parameter DURING (or before) the call. Thank you very much for supporting, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup after n seconds using originate ?
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=AllowCallerID
ActionID: AllowCallerID
Async: true
exten = _allow_caller_id.,1,Verbose(1,allow_caller_id gets ${EXTEN})
exten = _allow_caller_id.,n,Set(MyCallerID=${CUT(EXTEN,:,2)})
exten = _allow_caller_id.,n,GotoIf($[${LEN(${MyCallerID})}10]?NoCID)
exten = _allow_caller_id.,n,Set(CALLERID(num)=${MyCallerID})
exten = _allow_caller_id.,n(NoCID),Set(MyNumber=${CUT(EXTEN,:,3)})
exten = _allow_caller_id.,n,Set(MyTime=${CUT(EXTEN,:,4)})
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} calling ${MyNumber} for
${MyTime} seconds)
exten = _allow_caller_id.,n,Set(CALLERPRES()=allowed_not_screened)
exten = _allow_caller_id.,n,Dial${OutBoundDev}/${MyNumber},${MyTime},g)
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} call just got status
${DIALSTATUS})
exten = _allow_caller_id.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
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Re: [asterisk-users] Hangup after n seconds using originate ?
Here is how I do it, Mike
-- Perl Code --
my $phone_number=4918802;
my $testfile = /tmp/testin_$$.wav;
unlink $testfile;
my %resp = $astman-sendcommand( Action = 'Originate',
Channel =
DAHDI/$key/w$phone_number,
Variable = ARG1=$testfile,
Exten = 'SIP/170',
Context = 'testit',
ApplicationID = 1,
priority = 1,
Number = $phone_number
);
Context
[testit]
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,SetMusicOnHold(default)
exten = s,n,Waitexten(5,m)
exten = s,n,Verbose(record ${ARG1})
exten = s,n,record(${ARG1}|0|10|s)
exten = s,n,Waitexten(5,m)
exten = s,n,Goto(end-call|s|1)
Context 2
[end-call]
exten = s,1,Verbose(details - time ${DIALEDTIME} time2 ${ANSWEREDTIME}
status ${DIALSTATUS})
exten = s,n,AGI(clearorder.agi|${ABA}|${CHANNEL(language)})
exten = s,n,GotoIf($[${HANGUPCAUSE} = 0]?end-call|h|1)
exten = s,n,playback(vm-goodbye|noanswer)
exten = h,1,Hangup(${HANGUP_CAUSE})
This snippet calls 205-491-8802 (Telco Test line) and records 10 seconds of
tone into a file, then hangs up.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
mancyb...@gmail.com
Sent: Thursday, April 22, 2010 3:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something
like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
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Re: [asterisk-users] Hangup after n seconds using originate ?
Thanks for the comments, this did the trick :)
On Thu, 22 Apr 2010 13:51:35 -0700
Jim Dickenson dicken...@cfmc.com wrote:
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=AllowCallerID
ActionID: AllowCallerID
Async: true
exten = _allow_caller_id.,1,Verbose(1,allow_caller_id gets ${EXTEN})
exten = _allow_caller_id.,n,Set(MyCallerID=${CUT(EXTEN,:,2)})
exten = _allow_caller_id.,n,GotoIf($[${LEN(${MyCallerID})}10]?NoCID)
exten = _allow_caller_id.,n,Set(CALLERID(num)=${MyCallerID})
exten = _allow_caller_id.,n(NoCID),Set(MyNumber=${CUT(EXTEN,:,3)})
exten = _allow_caller_id.,n,Set(MyTime=${CUT(EXTEN,:,4)})
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} calling ${MyNumber} for
${MyTime} seconds)
exten = _allow_caller_id.,n,Set(CALLERPRES()=allowed_not_screened)
exten = _allow_caller_id.,n,Dial${OutBoundDev}/${MyNumber},${MyTime},g)
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} call just got status
${DIALSTATUS})
exten = _allow_caller_id.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
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Re: [asterisk-users] Hangup, SoftHangup
Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup, SoftHangup
Thanks Phillipp!, it works! Anahi Ludueña Date: Tue, 10 Nov 2009 14:44:09 +0100 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup, SoftHangup Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Sólo hay un loro experto en Windows 7 en todo el mundo. Y vive en Sietes ¡Cónocelo! http://www.sietesunpueblodeexpertos.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas da...@debsinc.com wrote:
So this *should* work??
[outgoing]
- exten = s,1,Dial(DAHDI/1/5551212,20)
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,Noop(you hung up)
- exten = h,2,Hangup
[incoming]
- exten = s,1,Answer
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,noop(you hung up)
- exten = h,2,hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Friday, October 23, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
if you are debugging visually then look at SIP BYE message ... who sent it
first
and on PRI who sent the DISCONNECT message first.
if you need to know that in the dialplan ... then if the originating
channel hanged up
then the dialplan should stop executing and go straight to h,1 even if
Dial(,,g) is used
also there is a channel variable HANGUPCAUSE and you can check what it
does on the next step
with Dial(,,g) and on h,1 ... since I don't know :)
Martin
On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH i...@saudihome.com wrote:
When Asterisk establish a call through an outbound trunk, Is there any way
I
can know who hang up the call first? The caller or the party called?
Thanks.
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Re: [asterisk-users] hangup from which side
That will work on an outgoing call. Apparently (AFAICS) there is no feature
in Answer to jump to H or continue like the Dial command has.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Tuesday, October 27, 2009 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas da...@debsinc.com wrote:
So this *should* work??
[outgoing]
- exten = s,1,Dial(DAHDI/1/5551212,20)
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,Noop(you hung up)
- exten = h,2,Hangup
[incoming]
- exten = s,1,Answer
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,noop(you hung up)
- exten = h,2,hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Friday, October 23, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
if you are debugging visually then look at SIP BYE message ... who sent it
first
and on PRI who sent the DISCONNECT message first.
if you need to know that in the dialplan ... then if the originating
channel hanged up
then the dialplan should stop executing and go straight to h,1 even if
Dial(,,g) is used
also there is a channel variable HANGUPCAUSE and you can check what it
does on the next step
with Dial(,,g) and on h,1 ... since I don't know :)
Martin
On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH i...@saudihome.com
wrote:
When Asterisk establish a call through an outbound trunk, Is there any
way
I
can know who hang up the call first? The caller or the party called?
Thanks.
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Re: [asterisk-users] hangup from which side
So this *should* work?? [outgoing] - exten = s,1,Dial(DAHDI/1/5551212,20) - exten = s,2,Noop(I hung up) - exten = s,3,Hangup - exten = h,1,Noop(you hung up) - exten = h,2,Hangup [incoming] - exten = s,1,Answer - exten = s,2,Noop(I hung up) - exten = s,3,Hangup - exten = h,1,noop(you hung up) - exten = h,2,hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Friday, October 23, 2009 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup from which side if you are debugging visually then look at SIP BYE message ... who sent it first and on PRI who sent the DISCONNECT message first. if you need to know that in the dialplan ... then if the originating channel hanged up then the dialplan should stop executing and go straight to h,1 even if Dial(,,g) is used also there is a channel variable HANGUPCAUSE and you can check what it does on the next step with Dial(,,g) and on h,1 ... since I don't know :) Martin On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH i...@saudihome.com wrote: When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb: When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? you could use the 'g' option of the Dial command together with some logic in the hangup extensions regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
We have queuemetrics and it does that
Here is some of the logic - (Obviously this wont work for you right out
of the box but you should be able to decipher the logic...)
[qm-queuedial]
; We use a global variable to pass values back from the answer-detect
macro.
; STATUS = U unanswered
;= A answered(plus CAUSECOMPLETE=C when callee hung up)
; The 'g' dial parameter must be used in order to track callee
disconnecting.
; Note that we'll be using the 'h' hook in any case to do the logging
when channels go down.
; We set the CDR(accountcode) for live monitoring by QM.
;
exten = s,1,NoOp,Outbound call - A:${QDIALER_AGENT}
N:${QDIALER_NUMBER} Q:${QDIALER_QUEUE} Ch:${QDIALER_CHANNEL}
exten = s,n,Set(CDR(accountcode)=QDIALAGI)
exten = s,n,Set(ST=${EPOCH})
exten = s,n,Set(GM=QDV-${QDIALER_AGENT})
exten = s,n,Set(GLOBAL(${GM})=U)
exten = s,n,Set(GLOBAL(${GM}ans)=0)
exten =
s,n,Macro(queuelog,${ST},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT},C
ALLOUTBOUND,-,${QDIALER_NUMBER})
exten =
s,n,Dial(${QDIALER_CHANNEL},300,gM(queuedial-answer^${UNIQUEID}^${GM}^${
QDIALER_QUEUE}^${QDIALER_AGENT}^${ST}))
exten = s,n,Set(CAUSECOMPLETE=${IF($[${DIALSTATUS} = ANSWER]?C)})
; Trapping call termination here
exten = h,1,NoOp( Call exiting: status ${GLOBAL(${GM})} answered at:
${GLOBAL(${GM}ans)} DS: ${DIALSTATUS} )
exten = h,n,Goto(case-${GLOBAL(${GM})})
exten = h,n,Hangup()
; Call unanswered
exten = h,n(case-U),Set(WT=$[${EPOCH} - ${ST}])
exten =
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},ABANDON,1,1,${WT})
exten = h,n,Hangup()
; call answered: agent/callee hung
exten = h,n(case-A)i,Set(COMPLETE=${IF($[${CAUSECOMPLETE} =
C]?COMPLETECALLER:COMPLETEAGENT)})
exten = h,n,Set(WT=$[${GLOBAL(${GM}ans)} - ${ST}])
exten = h,n,Set(CT=$[${EPOCH} - ${GLOBAL(${GM}ans)}])
exten =
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},${COMPLETE},${WT},${CT})
exten = h,n,Hangup()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus
Darilion
Sent: Friday, October 23, 2009 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb:
When Asterisk establish a call through an outbound trunk, Is there any
way I can know who hang up the call first? The caller or the party
called?
you could use the 'g' option of the Dial command together with some
logic in the hangup extensions
regards
klaus
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Re: [asterisk-users] hangup from which side
if you are debugging visually then look at SIP BYE message ... who sent it first and on PRI who sent the DISCONNECT message first. if you need to know that in the dialplan ... then if the originating channel hanged up then the dialplan should stop executing and go straight to h,1 even if Dial(,,g) is used also there is a channel variable HANGUPCAUSE and you can check what it does on the next step with Dial(,,g) and on h,1 ... since I don't know :) Martin On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH i...@saudihome.com wrote: When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup()-command does not hang up the line
I would try hanguponpolarityswitch=yes in my dadhi.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, May 12, 2009 3:09 PM To: Asterisk Mailing Subject: [asterisk-users] Hangup()-command does not hang up the line When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear silence, no special tone on the line that tells me the call was terminated by Asterisk, AND time keeps on counting on my cell phone as if the duration of the conversation continues. I see the following solution : - At the end of my context, I initiate the Congestion()-application to force the caller to hang up. But I think it must be enough just to call the Hangup()-command to make Asterisk terminate the conversation... But as I said : on my cell phone I see that time keeps on counting as if I'm still connected + no tone that the line was hung up. On the other hand : Asterisk detects the other end really good and registers when the caller has put down his phone and the conversation is terminated by the caller. Also a fax and a busy-tone is well detected. The option busydetect=yes is set in my chan_dahdi.conf... But this is not the problem. Is this a bug ?? Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup()-command does not hang up the line
On Tue, 12 May 2009, jonas kellens wrote: When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear silence, no special tone on the line that tells me the call was terminated by Asterisk, AND time keeps on counting on my cell phone as if the duration of the conversation continues. Replace the Asterisk box with a standard analogue phone and see what happens. I suspect you'll see the same. It happens in the UK too. The line will eventually clear, but it may take some time. I used to use it to transfer a call - ie. just put the phone back on-hook, then go to another phone and lift it... And you'll see it on old films where the bad guy phones a house and loads up the payphone with lots of money to stop the house being able to hang up the call and dial 999 ... Some exchanges do seem to clear the call much quicker now, but I think it's pot-luck, depending on the exchange and maybe hardware/software they have... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Detection After Originate (Asterisk Manager API)
On 24/04/2009 2:22 p.m., Saurabh Nirkhey wrote: I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my manager client . But if app2 is over and asterisk executes Hangup (app3),It never sends any packet to my client regarding Hangup of the call. I have given all permissions to manager user in manager.conf. Can somebody help me? Maybe use the UserEvent application before calling hangup: -= Info about application 'UserEvent' =- [Synopsis] Send an arbitrary event to the manager interface [Description] UserEvent(eventname[|body]): Sends an arbitrary event to the manager interface, with an optional body representing additional arguments. The body may be specified as a | delimeted list of headers. Each additional argument will be placed on a new line in the event. The format of the event will be: Event: UserEvent UserEvent: specified event name [body] If no body is specified, only Event and UserEvent headers will be present. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
Ok isn't this replacing a western hack with a bridge hack? The init 0 and init 6 probably aren't going to work anyway since (1) asterisk has to be running as root and (2) the path in * is limited if even existent, so the init command would work unless you had a copy or symlink in the asterisk directory. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, February 15, 2009 11:26 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup extensions via CLI? On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote: This will hang-up all channels even if multiples channels are open... Exten = _86,1,system(init 0) Use with Caution.? Only if Asterisk is running as root. Which is not recommended, anyway. And besides, I think you meant: Exten = _86,1,system(init 6) as we want to leave the extension available afterwards. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
On Tue, Feb 17, 2009 at 08:57:51AM -0600, Danny Nicholas wrote: Ok isn't this replacing a western hack with a bridge hack? The init 0 and init 6 probably aren't going to work anyway since (1) asterisk has to be running as root and I have already mentioned that this is a requirement. (2) the path in * is limited if even existent, so the init command would work unless you had a copy or symlink in the asterisk directory. # tr '\0' '\n' /proc/`cat /var/run/asterisk/asterisk.pid`/environ | grep ^PATH= PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin Init scripts tend to set the path explicitly. So those are just poor excuses for not using that fine hangup method. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Kindly consider the environment before printing this e-mail.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, February 13, 2009 3:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hangup extensions via CLI?
This version will hang up the given extension even if it has multiple channels
open:
asterisk -rx show channels | perl -lane print \asterisk -rx \'soft hangup
@F[0]\'\ if m.SIP/201. | bash
perl is always your friend when needing some programming mischief :)
l.
2009/2/12 Danny Nicholas da...@debsinc.com
Here's an improved hack to this bit of trickery:
Exten = _86,1,system('/usr/sbin/asterisk -rx soft hangup
$(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{
print $1 '} ))
Where dialing 861234 would hangup extension 1234
If this needs refinement, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30 (None) Up AppDial((Outgoing Line))
SIP/-09c59938 7...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' |
grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] Hangup extensions via CLI?
On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote: This will hang-up all channels even if multiples channels are open... Exten = _86,1,system(“init 0”) Use with Caution…☺ Only if Asterisk is running as root. Which is not recommended, anyway. And besides, I think you meant: Exten = _86,1,system(“init 6”) as we want to leave the extension available afterwards. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
| awk '{ print $1 '} )
what if there're also channels sip/201, sip/2011, sip/2012, sip/2013 et
al ?
--
Regards, /\_/\ All dogs go to heaven.
din...@alphaque.com(0 0) http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo The opinions here in no way reflect the opinions of my $a $b. |
| done; done |
+=+
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Re: [asterisk-users] Hangup extensions via CLI?
On Fri, Feb 13, 2009 at 06:08:45PM +0800, Dinesh Nair wrote:
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
| awk '{ print $1 '} )
Useless use of grep:
asterisk -rx soft hangup $(asterisk -rx 'show channels' | awk '/SIP\/201/
{print $1}' )
what if there're also channels sip/201, sip/2011, sip/2012, sip/2013 et
al ?
asterisk -rx soft hangup $(asterisk -rx 'show channels' | awk '/SIP\/201\/
{print $1}' )
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Hangup extensions via CLI?
This version will hang up the given extension even if it has multiple
channels open:
asterisk -rx show channels | perl -lane print \asterisk -rx \'soft
hangup @F[0]\'\ if m.SIP/201. | bash
perl is always your friend when needing some programming mischief :)
l.
2009/2/12 Danny Nicholas da...@debsinc.com
Here's an improved hack to this bit of trickery:
Exten = _86,1,system('/usr/sbin/asterisk -rx soft hangup
$(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{
print $1 '} ))
Where dialing 861234 would hangup extension 1234
If this needs refinement, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30(None) Up AppDial((Outgoing Line))
SIP/-09c599387...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' |
grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
|
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension
from
the system CLI. I understand this can probably be scripted using the
AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] Hangup extensions via CLI?
You guys think YOU'RE overdoing it... your solution works with a single line. My solution was some convoluted 100 line shell script! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Lenz Emilitri wrote: I have a feeling we're overdoing it :) l. 2009/2/12 Lukas Rypl r...@marconi.ttc.cz asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show message similar to Binary content: matched instead of expected line. It is necessary to use strings -a to filter output. So your example should be: asterisk -rx 'core show channels' | strings -a | grep SIP/7000 Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30(None) Up AppDial((Outgoing Line))
SIP/-09c599387...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] Hangup extensions via CLI?
Here's an improved hack to this bit of trickery:
Exten = _86,1,system('/usr/sbin/asterisk -rx soft hangup
$(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{
print $1 '} ))
Where dialing 861234 would hangup extension 1234
If this needs refinement, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30(None) Up AppDial((Outgoing Line))
SIP/-09c599387...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' |
grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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asterisk-users mailing list
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Re: [asterisk-users] Hangup extensions via CLI?
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show message similar to Binary content: matched instead of expected line. It is necessary to use strings -a to filter output. So your example should be: asterisk -rx 'core show channels' | strings -a | grep SIP/7000 Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
I have a feeling we're overdoing it :) l. 2009/2/12 Lukas Rypl r...@marconi.ttc.cz asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show message similar to Binary content: matched instead of expected line. It is necessary to use strings -a to filter output. So your example should be: asterisk -rx 'core show channels' | strings -a | grep SIP/7000 Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get the
channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI but
I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Hangup extensions via CLI?
Have you looked at soft hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Monday, February 09, 2009 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Hangup extensions via CLI? Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup issue
I've tried using a SIP client and when asterisk issue the Hangup function the SIP client indicate that the call is terminated. Maybe a SIP parameter with the pstn gateway ? Cyril SCETBON wrote: Hi guys, My asterisk server is connected to a pstn gateway using SIP. When I receive a call and use the Hangup command the pstn seems to not correctly see the request and the caller gets a 'number unknown message. Below are the debug message printed on the CLI : -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/192.168.19.1-0818f100, ) in new stack == Spawn extension (accueil, 483062608, 3) exited non-zero on 'SIP/192.168.19.1-0818f100' Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 384 ms (Method: ACK) set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 192.168.19.1, port 5060 Reliably Transmitting (NAT) to 192.168.19.1:53728: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 SIP/2.0 200 OK - --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK SIP/2.0 200 OK Any idea about what's happening and how to resolve it ? Regards -- Cyril SCETBON ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup conundrum with RxFAX
On Wed, 16 Apr 2008, lordfuknowsyou wrote: My thoughts now are to actually do a hangup at the end of the RxFAX and rely on a 'h' extension to pick it up and carry on with the 2nd half (which is PDFing and emailling the fax), but I'm concerned I'm going to lose the channel variables as it suggests on the wiki, so I'll lose the REMOTESTATIONID string and caller ID... Hi. Thats what I do and have not had a problem, we only do maybe 10-20 faxes a week though. I set my channel variables in a macro and then goto a context receivefax where I enter on s,1,Rx.Fax , on hangup I do the actual mailing and sending of the fax. Before the sending though I make sure the fax actually exists. Thanks for this. I'll give it a go! Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup conundrum with RxFAX
Gordon Henderson wrote: Heres something that's making me scratch my head... I'm using RxFAX on ISDN lines and in-general it's going well. However, there seems to be a case when the fax doesn't get delivered, but looking through the CDRs it seems that the call happened, RxFAX was executed .. time passed (1-2+ minutes) then hangup. I'm wondering if some FAX machines just hangup after the call rather than complete some sort of ending negotiation, or if the RxFAX part misses the end and just sees the hangup.. Now, in a normal fax machine, it's going to print the fax regardless, even if the last page is only half full because of a genuine line drop or hangup, but it seems that: [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the ... Returns -1 when the user hangs up. Returns 0 otherwise. So if it's returning -1, then the call/channel is hungup, and any dialplan instructions after it won't get executed, even though there might be some (or all) pages of the fax sitting in the receive file... Does this make sense to anyone, or am I barking up the wrong tree! My thoughts now are to actually do a hangup at the end of the RxFAX and rely on a 'h' extension to pick it up and carry on with the 2nd half (which is PDFing and emailling the fax), but I'm concerned I'm going to lose the channel variables as it suggests on the wiki, so I'll lose the REMOTESTATIONID string and caller ID... Anyone with any experience of this, or suggestions otherwise? Thanks, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi. Thats what I do and have not had a problem, we only do maybe 10-20 faxes a week though. I set my channel variables in a macro and then goto a context receivefax where I enter on s,1,Rx.Fax , on hangup I do the actual mailing and sending of the fax. Before the sending though I make sure the fax actually exists. hth Jeremy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Party
On Tue, 12 Dec 2006 15:27:06 +0200 Idris AVCI [EMAIL PROTECTED] wrote: Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. This is already written to the queue_log e.g. 1165572107|1165572085.354|french|Local/[EMAIL PROTECTED]|COMPLETEAGENT|20|2 or 1165495361|1165495218.23|french|SIP/1337-08234748|COMPLETECALLER|6|137 gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hangup Party
Thanks Gavin. We are not using built-in acd functions. Is there any way to report this in dialplan functions ? -Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 12, 2006 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hangup Party On Tue, 12 Dec 2006 15:27:06 +0200 Idris AVCI [EMAIL PROTECTED] wrote: Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. This is already written to the queue_log e.g. 1165572107|1165572085.354|french|Local/[EMAIL PROTECTED]|COMPLETEAGENT|20|2 or 1165495361|1165495218.23|french|SIP/1337-08234748|COMPLETECALLER|6|137 gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup or busy when the peer answer outgoing calls
Hi all!!, I haven't the 'r' options in the dial command. I also try to turn off busydetect and callprocess obtaining the same result.. If I turn off polarityswitch, I get hangup instead busy... The peer isn't busy because I'm trying with my movil phone, and whit known automatic operators from my telephony provider... when they answer my call, asterisk hangup the call.. Regards.. El mar, 10-10-2006 a las 14:39 -0800, Mojo with Horan Company, LLC escribió: If your Dial() cmd has an 'r' in the options, could it be that the ringing you're hearing is asterisk-generated, and the remote side actually is busy? Have you tried turning busydetect=no in zapata.conf? Moj Eloy Gomez wrote: Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: - loadzone = es defaultzone=es fxsks=1 zapata.conf -- [channels] signalling=fxs_ks busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes progzone=es context = contexto group = 1 channel = 1 And this is the asterisk log: -- Executing Dial(SIP/200-4803, ZAP/1/966736800|90) in new stack -- Called 1/966736800 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing PlayTones(SIP/200-4803, busy) in new stack -- Executing Wait(SIP/200-4803, 10) in new stack == Spawn extension (indeos, 0966736800, 103) exited non-zero on 'SIP/200-4803' Thanks all Eloy. -- Indeos Consultoria Eloy Gomez ([EMAIL PROTECTED]) Tel: 966787431 www.indeos.es ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup or busy when the peer answer outgoing calls
If your Dial() cmd has an 'r' in the options, could it be that the ringing you're hearing is asterisk-generated, and the remote side actually is busy? Have you tried turning busydetect=no in zapata.conf? Moj Eloy Gomez wrote: Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: - loadzone = es defaultzone=es fxsks=1 zapata.conf -- [channels] signalling=fxs_ks busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes progzone=es context = contexto group = 1 channel = 1 And this is the asterisk log: -- Executing Dial(SIP/200-4803, ZAP/1/966736800|90) in new stack -- Called 1/966736800 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing PlayTones(SIP/200-4803, busy) in new stack -- Executing Wait(SIP/200-4803, 10) in new stack == Spawn extension (indeos, 0966736800, 103) exited non-zero on 'SIP/200-4803' Thanks all Eloy. -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup on Panasonic KX-TEM824
No way if you are using fxs on panasonic and fxo on *. jorge [EMAIL PROTECTED] wrote: I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect HANGUP from this. Can anyone help me to get it work. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
http://www.voip-info.org/wiki/index.php?page=Australia%20Asterisk% 20Details Stumbled across this Reverse On Idle Condition (ROIC) 'feature' that sounds very promising. Will get it enabled later today and give it a go. On Tue, 2006-06-20 at 23:35 +1000, Carey O'Shea wrote: Well I've found out what was causing my duplicate logging: it was entirely a NAT issue. Found out it was only happening on some remote endpoints (and not all of them), and that different routers proved to not have duplicate logging. What part of NAT could cause this? Was it really sending all packets twice, or something like that? Just seems kinda strange. Anyway, it's no longer a problem. My original problem, however, remains. Phone doesn't stop ringing when it's meant to. Only happens when call is via my ZapATA. Any ideas/help/input is appreciated! Regards, Carey. On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
I'm having the exact same problem. Please any ideas? My IP phones keep ringing after PSTN hangup or PSTN answer... for about 6 or 7 seconds.On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show stopper... imagine ringing a business and beinganswered by 3 different people, one after the other, all talking overthe top of each other.On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything.
Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line?exten = s,1,Dial(IAX2/carey)Nothing else? Try adding a Hangup command on the next priority and see if that helps any.exten = s,2,HangupIf you already have a Hangup command in there,
then I apologize for wasting your time. :)Undrhil--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the
call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas
as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial("Zap/4-1", "IAX2/carey") in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial("Zap/4-1", "IAX2/carey") in new stack
-- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw)-- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup
'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show stopper... imagine ringing a business and beinganswered by 3 different people, one after the other, all talking overthe top of each other.On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea
wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the
loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line?exten = s,1,Dial(IAX2/carey)Nothing else? Try adding a Hangup command on the next priority and see if that helps any.exten = s,2,HangupIf you already have a Hangup command in there, then I apologize for wasting your time. :)Undrhil--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
Does anyone have any ideas as to what can cause this large delay to stop ringing? It's quite a show stopper... imagine ringing a business and being answered by 3 different people, one after the other, all talking over the top of each other. On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup extension
Thomas Kenyon wrote: I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten = h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as the call is being placed from. Strangely If I remove the outgoingdial macro and put it directly into the context instead, everything is fine (it doesn't ignore the hangup). Only problem now is that the userevents in macro-tapi don't appear to do anything. (still states that it is waiting for the phone to answer). I notice the Userevents that aren't doing anything open with a different channel than the others (gradwell-5 rather than Office-4). IS this relevant? == Manager 'tom' logged on from 192.168.0.8 -- Call accepted by 192.168.0.17 (format g729) -- Format for call is g729 Channel IAX2/Office-4 was answered. -- Executing AGI(IAX2/Office-4, setch.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/setch.agi setch.agi: IAX2/Office-4 setch.agi: IAX2/Office -- AGI Script setch.agi completed, returning 0 -- Executing UserEvent(IAX2/Office-4, TAPI|TAPIEVENT: LINE_NEWCALL IAX2/Office) in new stack -- Executing UserEvent(IAX2/Office-4, TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_DIALTONE) in new stack -- Executing UserEvent(IAX2/Office-4, TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_DIALING) in new stack -- Executing UserEvent(IAX2/Office-4, TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_PROCEEDING) in new stack -- Executing Dial(IAX2/Office-4, IAX2/gradwell/0xx|30|tM(tapi^1149870710.0|IAX2/Office)) in new stack -- Called gradwell/0xx -- Call accepted by 193.111.200.135 (format g729) -- Format for call is g729 -- IAX2/gradwell-5 is making progress passing it to IAX2/Office-4 -- IAX2/gradwell-5 answered IAX2/Office-4 -- Executing UserEvent(IAX2/gradwell-5, TAPI|TAPIEVENT: [~1149870710.0IAX2/Office] LINE_CALLSTATE LINECALLSTATE_CONNECTED) in new stack -- Executing UserEvent(IAX2/gradwell-5, TAPI|TAPIEVENT: [~1149870710.0!IAX2/Office] LINE_CALLSTATE LINECALLSTATE_HANGUP) in new stack -- Attempting native bridge of IAX2/Office-4 and IAX2/gradwell-5 -- Hungup 'IAX2/gradwell-5' == Spawn extension (astertapi, 0x, 6) exited non-zero on 'IAX2/Office-4' -- Executing UserEvent(IAX2/Office-4, TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) in new stack -- Hungup 'IAX2/Office-4' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup lag causing the answering of already answered calls
Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup issues
Hello, Load wctdm with debug=1, i.e, add this line to /etc/modprobe.conf: options wctdm debug=1 Then watch /var/log/messages (tail -f /var/log/messages will do it), and check when you are getting the first polarity reversal, you should get it before the first RING. If it happens that you get it when asterisk answers, that would explain your problem. BTW, is it a pstn line? or a gsm fct? If the later, you need to set it up for proper hangup detection in asterisk. Julian J. M. On 3/7/06, Carlos Prieto [EMAIL PROTECTED] wrote: Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=no I've tested with different values por busydetect set t yes and several busycount values. I'm using Asterisk 1.2.4 and Zaptel 1.2.3 with a Digium TDM400P with 2 FXO modules and Kewl Start Signalling. Thanks in advance for the help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup Detection (revisited)
Darrick Hartman wrote: A little background. I'm integrating asterisk as the voicemail service for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2 is used to connect an analog device (such as a voice modem) to the pbx. In the past we've used vgetty and a voice modem with varying degrees of success. If you haven't yet, I'd turn on busydetect in zapata.conf. Can't hurt and might (although unlikely) work (I had to turn it on to make it work on my system). Switching to loop-start might be worth a try, too. For a while my VM * system wasn't doing disconnect detection, and it was OK. I had trouble with the single-port cheapo cards off eBay with the silence thresholds, but using a TDM400P card fixed that for me. Also make sure all you menus will time out and hang up. You could try posting to the Nortel list: http://www.tgrace.com/mailman/listinfo/nortel-list They've been very helpful and kind to me. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: Re: [Asterisk-Users] hangup detection
Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded.This hasn't worked though, asterisk doesn't seem to detect the busy signal.Does asterisk require a the signal to be in a certain power range? The signal I getis very quiet.Thanks for your helpRegardsJonathanOn Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote: Hi everybody! Jonathan wrote: Hi, I'm using a td400p card with an FXO port and asterisk 1.2.1 in South Korea and asterisk isn't detecting when PSTN callers hangup. I've gone through all the settings related to hangup detection and none work. I've tried: hanguponpolarityswitch=yes callprogress=yes busydetect=yes busycount=6 I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in Colombia and tried with a lof of loadzone= Debug doesn't show reverse polarity events so I'm pretty stuck. I've got zaptel configured with a loadzone of US and kewlstart signialling. Has anybody had success with these cards/asterisk in South Korea? ?Or in the world? We implemented a busypattern= option for the zapata.conf that might help you. Test like so: Dial into your Asterisk system via the FXO port to an extension on your box. Now hang up from the outside. Listen to the call on the internal extension. If you hear a regular beep-beep tone of some sort, busypattern= might help you. You need to time exactly the length of the beep and the length of the silence. (To get it nice and accurate, record it, then load into Audacity and measure). Say it comes out at 750 msec of beep, 500 msec of silence. Then adjust your zapata.conf like so: busydetect=yes callprogress=no busypattern=750,500 busycount=4 Regards, Steve Davies _ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
On Tue, 10 Jan 2006, [EMAIL PROTECTED] wrote: Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan Yeah - it needs to be reasonably loud to be detected. Too bad. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote: Hi everybody! Jonathan wrote: Hi, I'm using a td400p card with an FXO port and asterisk 1.2.1 in South Korea and asterisk isn't detecting when PSTN callers hangup. I've gone through all the settings related to hangup detection and none work. I've tried: hanguponpolarityswitch=yes callprogress=yes busydetect=yes busycount=6 I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in Colombia and tried with a lof of loadzone= Debug doesn't show reverse polarity events so I'm pretty stuck. I've got zaptel configured with a loadzone of US and kewlstart signialling. Has anybody had success with these cards/asterisk in South Korea? ?Or in the world? We implemented a busypattern= option for the zapata.conf that might help you. Test like so: Dial into your Asterisk system via the FXO port to an extension on your box. Now hang up from the outside. Listen to the call on the internal extension. If you hear a regular beep-beep tone of some sort, busypattern= might help you. You need to time exactly the length of the beep and the length of the silence. (To get it nice and accurate, record it, then load into Audacity and measure). Say it comes out at 750 msec of beep, 500 msec of silence. Then adjust your zapata.conf like so: busydetect=yes callprogress=no busypattern=750,500 busycount=4 Regards, Steve Davies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Hi everybody! Jonathan wrote: Hi, I'm using a td400p card with an FXO port and asterisk 1.2.1 in South Korea and asterisk isn't detecting when PSTN callers hangup. I've gone through all the settings related to hangup detection and none work. I've tried: hanguponpolarityswitch=yes callprogress=yes busydetect=yes busycount=6 I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in Colombia and tried with a lof of loadzone= Debug doesn't show reverse polarity events so I'm pretty stuck. I've got zaptel configured with a loadzone of US and kewlstart signialling. Has anybody had success with these cards/asterisk in South Korea? ¿Or in the world? Thanks JC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection - TDM400P
hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection - TDM400P
Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection - TDM400P
Here in Spain we had that problem since the hangup here is done by changing line polarity. It is solved by aplying this patch: http://www.maxosystem.net/asterisk/asterisk-stable-polarity-v5.diff $ cd /usr/src/asterisk/channels $ patch chan_zap.c /your/route/here/asterisk-stable-polarity-v5.diffand in zapata.conf :answeronpolarityswitch=yes hanguponpolarityswitch=yesHope it helps ;)- Original Message - From: Marco Supino [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 17, 2005 5:20 PM Subject: Re: [Asterisk-Users] Hangup detection - TDM400P Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup problem
On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara wrote: i have a box running debian sarge with asterisk installed from distribution packages: CLI show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small office with just a few phones) I have one Zap (PSTN) line connected to it and one SIP to a local provider. After some googling most things seem to work well, but i'm having problems with Hangup. This affects both the Zap interface and the SIP connection to the provider. No matter who tries to hang up an established call, its not properly finished. In such case, with the other node just silence is heard (and not congestion/busy signal). This includes calls initiated both from outside and from inside with a voip phone connected to the asterisk. on asterisk console i'm getting these messages: == Spawn extension (incoming, s, 2) exited non-zero on 'SIP/1012082-8408' == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/1012082-8408' here is the relevant part of the dialplan: [ Context 'incoming' created by 'pbx_config' ] 'h' =1. Hangup() [pbx_config] 's' =1. Ringing() [pbx_config] 2. Dial(SIP/11|5) [pbx_config] 3. Dial(SIP/11SIP/21|20) [pbx_config] You dial. After a hangup you attempt to dial again. Right? What do you try to do there? Ring the one that is free? 5. Hangup() [pbx_config] 't' =1. Hangup() [pbx_config] -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup Faster
David Sampson wrote: Hello My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesnt disconnect right away. Any ideas on how to resolve this? Thanks, Dave In zapata.conf put this line. rxflash=50 This may prevent flash from working properly. I don't know for certain because I have those functions turned off. threewaycalling=no transfer=no ___ 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] Hangup detection on Panasonic KXTD816
Hilton Williams wrote: Hi I have a Digium TDM400 card with 4 FXO modules connected to the extension ports on a Panasonic KXTD816. I'm using [EMAIL PROTECTED] v1.0, which has Asterisk 1.07. There's a problem that Asterisk doesn't detect when the line is disconnected on the Panasonic. The Panasonic doesn't provide polarity reversal or current drop or anything like that to indicate hangup. It just plays the dial tone again. Correct. When I have to interface with a PBX I use FXS ports on Asterisk connected to the FXO/CO ports of the PBX. This seems to (mostly) work well, since PBXs tend to be MUCH better at figureing out that a line is disconnected than Asterisk is. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Hangup Issues on TDM40B FXO Australia
I have a similar issue. I have 2 pstn lines and a phone plugged into my tdm400. If I make a call to the outside using the phone, and the pstn number is engaged, and I hang up, the line is not freed. I have been restarting asterisk to get my external line back. This does not happen if I make the same call from my pc (using sj phone). Malcolm [EMAIL PROTECTED] wrote: Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 ___ 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] Hangup Issues on TDM40B FXO Australia
Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 23 May 2005 12:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn -- -- This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. -- -- ___ 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] Hangup Issues on TDM40B FXO Australia
I had a similar issue both with the X100P clones and TDM400. Both were fixed by enabling AU zone and the busydetect functions. Don't forget a full asterisk reload needs to take place after changing Zap conf files, not just a soft-reload. Best way is to reboot the computer. Mike I have a similar issue. I have 2 pstn lines and a phone plugged into my tdm400. If I make a call to the outside using the phone, and the pstn number is engaged, and I hang up, the line is not freed. I have been restarting asterisk to get my external line back. This does not happen if I make the same call from my pc (using sj phone). Malcolm [EMAIL PROTECTED] wrote: Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 ___ 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] Hangup Issues on TDM40B FXO Australia
Thanks for the response all I'm Currently running version 1.0.7 of asterisk, Key configuration lines as below, for those who have it working is there anything that stands out as incorrect? * Zaptel.conf fxsls=1-4 loadzone=au defaultzone=au *** Zapata.conf signalling=fxs_ls switchtype=national *** zapata-channels.conf signalling=fxs_ls *** zapata_additional.conf ;;[EXT] signalling=fxo_ls echotraining=400 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=yes busydetect=yes busycount=3 channel=1 *** -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Monday, 23 May 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 23 May 2005 12:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn -- -- This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. -- -- ___ 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 This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit
RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
Thanks Terry noticed [EMAIL PROTECTED] 0.7 will try version 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Monday, 23 May 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Hi, I have 2 Asterisk servers in .pg and 2 in .au In .pg I have had to configure them as if they were in .au and use LS signaling. I am using the latest Asterisk @ Home (1.0) and it is working well with 1 TDM400P for interfacing with the PSTN lines. Previously I had exactly the problem you have described using Asterisk @ Home (0.7). I also had a memory leak problem in that all the memory 512Mb would be gradually used up and after about 3 days the audio would begin to suffer. The upgrade was a full reinstall and rebuild from documentation and once completed, the problems have not reappeared. These server are also NTP servers and DHCP servers Regards, T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 23 May 2005 12:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn -- -- This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. -- -- ___ 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 This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ 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
