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
[asterisk-users] Hangup Reason
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? Thanks all, Alex -- _ -- 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
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Hangup() not working for handsets using pls transport?
Hi all,
I managed to get tls transport going with asterisk 16.14.0, and set one
handset (SOFTPHONE_B) to use the transport. I have set up a few other
handsets (both soft and hard) to use udp and tcp transports:
voip1*CLI> pjsip show endpoints
Endpoint:
I/OAuth:
Aor:
Contact:
Transport:
Identify:
Match:
Channel:
Exten: CLCID:
==
Endpoint: f30A0A01 Not in
use0 of inf
InAuth: f30A0A01/f30A0A01
Aor: f30A0A01 1
Contact: f30A0A01/sip:f30A0A01@128.171.77.1 4800418965
NonQual nan
Transport: transport-udp udp 0 0 0.0.0.0:5060
Endpoint: f30B0B02 Not in
use0 of inf
InAuth: f30B0B02/f30B0B02
Aor: f30B0B02 1
Contact: f30B0B02/sip:f30B0B02@128.171.77.4 615cc2a2c6
NonQual nan
Transport: transport-udp udp 0 0 0.0.0.0:5060
Endpoint: SOFTPHONE_A
Unavailable 0 of inf
InAuth: SOFTPHONE_A/SOFTPHONE_A
Aor: SOFTPHONE_A2
Transport: transport-tcp tcp 0 0 0.0.0.0:5060
Endpoint: SOFTPHONE_B Not in
use0 of inf
InAuth: SOFTPHONE_B/SOFTPHONE_B
Aor: SOFTPHONE_B2
Contact: SOFTPHONE_B/sip:SOFTPHONE_B@128.171.168.23 78257ab30a
NonQual nan
Transport: transport-tls tls 0 0 0.0.0.0:5061
Objects found: 4
voip1*CLI>
For testing, I have the following in /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()
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)
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)=${bar})
> same => n,Return()
>
> [default]
> exten => _X.,1,NoOp(New test call)
> same => n,Set(CHANNEL(hangup_handler_push)=sub-hanguphandler,s,1)
> same => n,Dial(...)
> same => n,Hangup()
>
>
> With the above config:
>
> 1- An answered call: foo is filled in db
> 2- A cancelled call: foo is filled in db
> 3- A failed call: foo is NOT filled in db
>
> In all 3 cases, with verbose logs enabled I can see
> the sub-hanguphandler subroutine is being executed, so it's confusing.
>
> I would understand if the hangup handler is NOT executed for failed calls,
> but seeing it in the logs and then seeing the field empty in the db doesn't
> make sense to me.
>
> Any tips on where/how I can troubleshoot this?
>
>
> Thanks,
> Joel.
>
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[asterisk-users] Hangup-handler on failed calls
Hello,
I have a setup with asterisk 16.8.0, I'm facing a problem where calls that
fail (CONGESTION) don't have filled in some extra fields we add to the CDRs
in the database.
We use cdr_adaptive_odbc with MySQL as backend.
To simplify the scenario:
[sub-hanguphandler]
exten => s,1,Set(CDR(foo)=${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
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Hangup hook to put back a call into a queue
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
Return
MacroExit
how to redirect the call to the queue?:
Queue(1000)
ChannelRedirect(${CHANNEL},,1000,1)
Gosub(ext-intercom,*801000,1())
dial-one,HhTtrM(auto-blkvm),1000
and what is the reason I can’t put the call back to the queue?
I know that I'm already in the hangup sequence, but still wouldn't like
to hangup.
or this can't be done in the hangup handler?
thank you for your help in advance.
regards.
--
Levente "Si vis pacem para bellum!"
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Re: [asterisk-users] Hangup handler gosub error with asterisk 16.4.0.
On 6/1/19 9:18 AM, Harley Peters wrote:
I am receiving the following errors on any hangup handler subroutines.
[2019-05-31 18:22:13.958] VERBOSE[23943][C-0009] app_stack.c:
PJSIP/104090401-000a Internal Gosub(PreventForwardingLoop,s,1)) start
[2019-05-31 18:22:13.958] NOTICE[23943][C-0009] pbx.c: No such label
'1)' in extension 's' in context 'PreventForwardingLoop'
[2019-05-31 18:22:13.958] WARNING[23943][C-0009] pbx.c: Priority
'1)' must be a number > 0, or valid label
[2019-05-31 18:22:13.958] ERROR[23943][C-0009] app_stack.c: Gosub
address is invalid: 'PreventForwardingLoop,s,1)'
Dialplan:
exten =>
_1NXXNXX,n,Set(CHANNEL(hangup_handler_push)=PreventForwardingLoop,s,1))
[PreventForwardingLoop]
exten =>
s,1,Set(DELETEKEY=${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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[asterisk-users] Hangup handler gosub error with asterisk 16.4.0.
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
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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
[asterisk-users] hangup the _called_ channel ?
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. 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() How do I hangup the called channel, and leave the calling channel listening to the file ? -- _ -- 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
[asterisk-users] hangup handlers & unwanted cdr
hi, i'm using hangup handlers on Asterisk13 with standard answered calls i have 1 CDR per call with scenario call from voip->mobile, call rejected on mobile i have 2 CDRs i dont want the second CDR without hangup handlers i have 1 CDR do you think its bug or its feature of hangup handlers? *** 1. row *** calldate: 2017-05-31 15:50:28 clid: "voip_number" src: voip_number dst: mobile_number dcontext: route_phones_1 channel: SIP/vr1a915-001e dstchannel: SIP/siptrunk-001f channtype: lastapp: Dial lastdata: SIP/siptrunk/mobile_number,120,tTb(pre_dial_handler^callee^1)B(pre_dial_handler^cal duration: 10 billsec: 0 disposition: NO ANSWER amaflags: 3 accountcode: uniqueid: 1496238628.30 hangupcause: stamp: 2017-05-31 15:50:38 linkedid: 1496238628.30 sequence: 30 *** 2. row *** calldate: 2017-05-31 15:50:28 clid: "mobile_number" src: mobile_number dst: mobile_number dcontext: trunk_context_1 channel: SIP/siptrunk-001f dstchannel: channtype: lastapp: Return lastdata: duration: 9 billsec: 0 disposition: FAILED amaflags: 3 accountcode: uniqueid: 1496238628.31 hangupcause: stamp: 2017-05-31 15:50:38 linkedid: 1496238628.30 sequence: 31 -- _ -- 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
[asterisk-users] hangup locked channels
Hello, When I run "core show channels verbose", I seen around 10 locked channels that are lasting hundreds of hours (I haven't restarted Asterisk for more than 7 weeks). I try to hangup those channels using "hangup request ", but nothing happens. What could I do (besides restarting Asterisk)? Thanks Dov -- _ -- 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 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
[asterisk-users] hangup call gw FXO
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 -- 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 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
[asterisk-users] Hangup Chanel when a peer unregisters
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 -- _ -- 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
[asterisk-users] Hangup check during long running macro called by M option on Dial
I have built a dialplan which dial to someone with option M. Dial (SIP/1000,,M(MYMACRO)) Both parties are SIP phones. MYMACRO expects person on SIP/1000 dial 5 (using read) then exits - and doing so it bridges my phone (SIP/2000) with SIP/1000. If SIP/1000 hangs up before dial 5 - ok the call ends. if SIP/2000 hangs up before SIP/1000 dial 5 - the macro is unaware and keeps waiting SIP/1000 dial out 5. When it occurs the call ends. In the meanwhile I got the following warning: WARNING[4486]: chan_sip.c:4210 __sip_autodestruct: Autodestruct on dialog ' 1826448978@192.168.2.121' with owner SIP/2000-0014 in place (Method: BYE). Rescheduling destruction for 1 ms I am using Asterisk 1.8 Thanks Valter -- _ -- 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
[asterisk-users] Hangup cause 111 after call pickup
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///sipacc//3-147c, 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. Kind regards, Jonas. -- _ -- 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
[asterisk-users] Hangup problems
Hello!! I have asterisk 1.6.2.10 and whenever there are more than 60 calls queued the following problem ocurrs. The agents hangup the calls but the do not receive new calls for some seconds or even minutes. If I seek throughout the full log I encounter that the Bye message coming from asterisk to the agents failes to arrive. If I make calls between extensions what I see is the following: Extension A calls extension B. Extension A hangups. Bye message is send from extension A to asterisk. Asterisk sends an Ok message to extension A. After some seconds or even minutes, extension B receives Bye message from asterisk. Any help will be appreciated!! Also I see messages like: Line 71344: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: Scheduling destruction of SIP dialog '065e1e5e612fcbf947d4f9044dd8c...@xxx.xxx.xxx.xxx'in 6464 ms (Method: INVITE) Line 71347: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination: Parsing sip:2...@xxx.xxx.xxx.xxx:XXX;rinstance=b877e4c23d44a205 for address/port to send to Line 71348: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination: set destination to XXX.XXX.XXX.XXX, port Line 71349: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:XXX: Any body can tell m what these Scheduling destruction of SIP dialog messages mean? Thanks!!! -- _ -- 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
[asterisk-users] Hangup not detected
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? Thanks :-) Anam. -- _ -- 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
[asterisk-users] hangup not detected?
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:
[AAA_27_EMP]
exten = s,1,Answer
same = n,Set(CDR(accountcode)=27_EMP)
same = n,Set(comp_num=27)
same = n,Set(readprompt=AAA/enter_employee_number)
same = n,Set(truck_text=employee number)
same = n,Set(validate_func=AAA_VALIDATE_EMP_NUM)
same = n,Set(get_param1=27)
same = n,Set(get_param2=E)
same = n,Set(read_length=7)
same = n,Goto(DB_LOOKUP,s,1)
[DB_LOOKUP]
exten = s,1,NoOp()
same = n(getid),Read(account_id,${readprompt},${read_length},,3,5)
same = n,Gotoif($[ ${LEN(${account_id})} = 0]?timeout_hangup)
same = n(validateid),Verbose(validating id ${account_id})
same = n,Set(CDR(userfield)=${account_id})
same = n,GotoIf($[${account_id}==*]?AAACompMenu,s,1)
same =
n,Set(ID_VALIDATED=${validate_func}(${get_param1},${account_id}))
same = n,GotoIf($[${ID_VALIDATED}==0]?badid)
same = n(goodid),Verbose(getting schedule for id ${account_id}
AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})))
same =
n,Set(ODBC_ID=${AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})})
same = n,GotoIf($[${ODBCROWS} 1]?no_schedule)
same = n,Verbose(odbcrows count: ${ODBCROWS})
same = n,Set(COUNTER=1)
same = n,Set(AAA_OUTPUT=Schedule for ${truck_text} ${account_id}: )
same = n,While($[${COUNTER} = ${ODBCROWS}])
same = n,Set(ARRAY(id,data)=${ODBC_FETCH(${ODBC_ID})})
same = n,Set(AAA_OUTPUT=${AAA_OUTPUT}${data}. )
;same = n,Swift(${data})
same = n,Set(COUNTER=$[${COUNTER} + 1])
same = n,EndWhile()
same = n,ODBCFinish()
same = n,NoOp(${get_param2})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, S, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Set(MAX_REPEAT=5)
same = n(readschedule),Swift(${AAA_OUTPUT})
same = n,Set(MAX_REPEAT=$[${MAX_REPEAT}-1])
same = n,Gotoif($[${MAX_REPEAT} = 0]?timeout_hangup)
same = n,Read(return_id,AAA/end_of_schedule,${read_length},,,2)
same = n,Gotoif($[ ${LEN(${return_id})} = 0]?readschedule)
same = n,Set(account_id=${return_id})
same = n,Goto(validateid)
same = n(timeout_hangup),Swift(No ${truck_text} entered. Goodbye)
same = n,Hangup()
same = n(badid),Set(AAA_OUTPUT=Invalid ${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, I, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
same = n(no_schedule),Set(AAA_OUTPUT=No schedule found for
${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, N, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
Thanks in advance
-Justin
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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
[asterisk-users] hangup problem on T1 span
Hello all, I'm trying to solve a problem on a T1 span setup wherein calls are apparently not hanging up properly. 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?). 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. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 -- _ -- 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
[asterisk-users] Hangup Cause and SIP Response Code
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?
Thanks
Bryant
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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
[asterisk-users] hangup delayed very much on fastagi appliaction of asterisk 1.6
Hello my friends, Previously, we developed fastagi application with Erlang to run on asterisk 1.4, it run very well. But when we try to migrate this application to interface with asterisk 1.6.2.13( the same with 1.6.1 or 1.6.0), We found the hangup issue there, the hangup event will delayed a few minutes when callee hangup , or make an hangup request on cli. The channel be hungup don't release at once on hangup action performed. And on AMI , there is also no hangup event output at once. So my fastagi application hold there too. It'll delay a few minutes until the hangup event output from cli or AMI.we tried dial over dahdi and sip channel, the result is same. This issue don't happen on 1.4, has anyone here experienced the same issue? And share the resolving experiences here ? Is there any special configuration to control this hangup delay behavior? Thank you in advance! Best Regards, Thomas Liu image001.gif-- _ -- 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
[asterisk-users] Hangup Detection Problem In Turkey
Hi, Although zonedata.c contains ITU E.180 recommendations for Turkey, we are still experiencing unrecognized hangups from Turk Telekom PSTN lines when callers hangup. Turk Telekom does *not* provide supervised disconnects on analog PSTN, and the tone we receive we when caller hangs up is similar to busy, with three short beeps, followed by one long beep, which keeps repeating. We've tried busydetect, polarityswitch, etc. with no success. As it stands, asterisk using dahdi (with Digium TDM410p) does not work in Turkey. Can offer small bounty for any developer wishing to solve this issue. The technology we have used shown below; Asterisk 1.6.2.8 Libpri 1.4.11.1 Dahdi 2.3.0 Ubuntu Server 9.10 Kind Regards -- _ -- 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
[asterisk-users] Hangup Detection
Is there any way, i can detect in asterisk that which party hanged up the call either from A side or B. Both parties are using SIP protocol. I am using Asterisk 1.4.27 Shariq Khan 0333-3501125 -- _ -- 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
[asterisk-users] Hangup after n seconds using originate ?
Hi All, I would like to know if you can confirm that, if using origination via AMI, as documented here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate it is not possible to set the max duration of a call. I mean: what you would do with the L (limit) parameter of the command Dial, is not possible when originating. As well as using the absolute timeout, as documented here: http://www.asteriskguru.com/tutorials/timeoutabsolute_function.html can't be done when originating. Is this true ? I'm using version 1.4. Thanks for supporting, have a nice day. 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 ?
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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[asterisk-users] Hangup after 1 second of ringing ?
Hi All, does the Asterisk's 'Dial' command have some hooks to execute commands as soon as the 'ringing' signal is received ? For example: can a call be dropped 1 second after the called party's phone started to ring ? I'm using version 1.4. Thanks for supporting, have a nice day. 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
[asterisk-users] Hangup
Hi, 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? 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
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
[asterisk-users] hangup from which side
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
[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
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
[asterisk-users] Hangup Detection After Originate (Asterisk Manager API)
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? Thanks Regards === (-: Saurabh :-) === French is the language of love,For everything else there is 'C' Every search begins with beginner's luck and ends with the victor being severly tested -Paulo Coehlo ___ -- 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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asterisk-users mailing list
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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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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] 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
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
[asterisk-users] hangup problem(for spa400)
Hi all, I have asterisk connected to my voice application server. Asterisk is connected and registering to a linksys spa400 box. I am running an application on a perticular extention (141). Here is a snip from my extensions.conf... exten = spa400,s,MyApp(/etc/asterisk/MyAppConfig.conf) exten = spa400,s+1,Hangup when an incoming call comes,It is accepted properly,And the application executes successfully,Also I see SIP BYE going to spa400 and getting a 200 Ok for BYE from spa400.But the problem is that the call is not really disconnected,Caller's billing doesn't stop, It takes atleast a minute,before the 'call end tone' can be heard by the caller. It can be a spa400 issue,(As I can see a SIP BYE at asterisk end). Seeking for some help/pointers Thanks -- === (-: Saurabh :-) === French is the language of love,For everything else there is 'C' Every search begins with beginner's luck and ends with the victor being severly tested -Paulo Coehlo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup?
I've got a unique situation and think it may be the lack of the Hangup command in the dialplan that is creating the issue.Can anyone elaborate on why it is, or is not, important to use hangup in the dialplan. Presently I don't have the first instance of it in my dialplan, however, I see some things in the debugging that might be cleaner if I did implement hangup I have approximately 140 extensions provisioned off this asterisk server and about 8 IVRs...So as you might expect, it is quite busy... Thanks, -Joe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup channel
Hi all, I am getting a weird error here. When i send a call to a sip peer on one of our servers i get a 'Nobody picked up in -1 ms' immediately following the SIP INVITE then the call hangs up. I do not have a timeout in the Dial, if i send the call to a different peer the call works fine. I am running 1.2 SVN 2006-02-22 Here is the dial statement used: Executing Dial(SIP/1ST LEG, SIP/2ND CALL LEG||t) in new stack __ Not happy with your email address?. Get the one you really want - millions of new email addresses available now at Yahoo! http://uk.docs.yahoo.com/ymail/new.html___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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
[asterisk-users] Hangup issue
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
