Re: [asterisk-users] call pickup
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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] call pickup
Search for dialog-info pickup -Original Message- From: Marek Cervenka cerv...@fpf.slu.cz Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 07 Oct 2011 09:47:45 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] call pickup On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some NOTIFY to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without monitoring the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- 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] Problem With Playing Busy Tone
Hi Since upgrading to 1.8.4.3 my callers no longer hear busy tone when I use playtones(). Here is the CLI output on such a case http://pastebin.com/TMBFhngh Any ideas anyone? Regards Jon -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- Daniel -- _ -- 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] Asterisk 1.8.7 and client outside network
Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? Thanks for your help -- Daniel -- _ -- 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] dahdi show status command not avilable in CLI
Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 # dahdi_scan [1] active=yes alarms=OK description=Wildcard X100P Board 1 name=WCFXO/0 manufacturer=Digium devicetype=Wildcard X100P location=PCI Bus 02 Slot 02 basechan=1 totchans=1 irq=193 type=analog port=1,FXO These outputs shows that the modules are loaded correctly. Any other clues ? Michael.k On Thu, Oct 6, 2011 at 8:43 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, if you type dah followed by TAB and nothing appears, it means you do not have dahdi module loaded or dahdi_cfg application not launched before starting asterisk. Giorgio On 10/06/2011 04:57 PM, michael k wrote: Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my
Re: [asterisk-users] dahdi show status command not avilable in CLI
It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 # dahdi_scan [1] active=yes alarms=OK description=Wildcard X100P Board 1
Re: [asterisk-users] dahdi show status command not avilable in CLI
Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) 1 channels to configure. Setting echocan for channel 1 to mg2 #
Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
remove the c argument Kristijan 2011/10/7 Administrator TOOTAI ad...@tootai.net: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- Daniel -- _ -- 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] Add SIP diversion header in originate from AMI?
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected
to an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to do this
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
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Re: [asterisk-users] Add SIP diversion header in originate from AMI?
Try run your outbound leg through a Local channel.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Oct 7, 2011, at 11:03 AM, Tobias Steen tobias.st...@s2.se wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected to
an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to do this
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
--
_
-- 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
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--
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http://www.asterisk.org/hello
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Re: [asterisk-users] Add SIP diversion header in originate from AMI?
You can dial a local channel which executes a dial plan that does what you want.
Channel: Local/dial_number@cfmc_cdi_private
This will use exten dial_number in the cfmc_cdi_private context.
If you add something like this to the originate packet
Variable: CfMC_Use_CID=5419712513
You can use ${CfMC_Use_CID} to get the value.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 7, 2011, at 8:03 AM, Tobias Steen wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected to
an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to dothis
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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[asterisk-users] DIDs in Singapore
Can anyone suggest an ITSP with Singapore DIDs and local Singapore termination? Cheers, j -- _ -- 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] dahdi show status command not avilable in CLI
Please paste the configurations in the #included files as well. On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote: Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ? Michael.k On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi Michael, what if you reload the module chan_dahdi from within the * CLI? It should give some hints. Giorgio On 10/06/2011 05:22 PM, michael k wrote: Hi Giorgio, Thanks for your reply. I will produce some output for your reference. # lsmod | grep dahdi dahdi_echocan_mg2 39688 1 dahdi_transcode42372 1 wctc4xxp dahdi_voicebus 79424 2 wctdm24xxp,wcte12xp dahdi 238384 14 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 35265 2 wctdm24xxp,dahdi # service dahdi status ### Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) 1 FXOFXSKS (SWEC: MG2) (battery) # dahdi_cfg -vv DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
On 10/07/2011 07:46 AM, Administrator TOOTAI wrote: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? You included the 'c' option to ReceiveFAX, telling it to act as the 'caller', even though it isn't the caller. This argument is parsed by ReceiveFAX in spite of it not being supported because the older app_fax version did support it, and we didn't want dialplans that included it to silently ignore the 'c' option. The same is true for the 'a' option; you'll note that neither of them are included in the documentation for the ReceiveFAX and SendFAX applications, and shouldn't be used. Why did you specify the 'c' option? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco AS5400XM
On Thu, Oct 6, 2011 at 11:25 AM, Kyle Sexton k...@mocker.org wrote: I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP signaling. Has anyone had any experience with these devices? The feature cards that Cisco sells can be a little confusing. I'm thinking something like below is what I need. (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem) (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card Any thoughts would be appreciated. Thanks. We also looked it one time. One of the model is no longer supported by cisco. The replacement model with all DSP loaded was quite expensive, about $20k per box even on ebay. -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 02:20 PM, James Sharp wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 Thanks for that; it helps. First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). If you would, please retry this with the HEAD of the Asterisk 10 branch instead of 10.0.0-beta1, and also capture the UDPTL packets themselves so we can see what they contained. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
for which user/number sip reinvite is for? ooh! you are running a direct application without any dialplan or user, may be that is the cause! I think you should first write fax dialplan, reload asterisk and test again with originate but this time with extension not application. Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Oct 8, 2011 at 12:20 AM, James Sharp ja...@fivecats.org wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 -- __**__**_ -- 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-**usershttp://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] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 03:06 PM, Nasir Iqbal wrote: for which user/number sip reinvite is for? ooh! you are running a direct application without any dialplan or user, may be that is the cause! I think you should first write fax dialplan, reload asterisk and test again with originate but this time with extension not application. No, none of that is relevant. It's perfectly acceptable to call SendFAX() on a CLI/AMI/spool-originated channel. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote: First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). Theres a few t30-nosignal packets at the beginning, but then they transition to other t30 packets, including CNG, CED, preambles, training and data. Wireshark says the sequence number is always 0, so it appears that Asterisk is not mis-displaying http://pastebin.ca/2087784 I can provide the raw tcpdump file if needed. If you would, please retry this with the HEAD of the Asterisk 10 branch instead of 10.0.0-beta1, and also capture the UDPTL packets themselves so we can see what they contained. -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 03:29 PM, James Sharp wrote: On 10/07/2011 04:04 PM, Kevin P. Fleming wrote: First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). Theres a few t30-nosignal packets at the beginning, but then they transition to other t30 packets, including CNG, CED, preambles, training and data. Wireshark says the sequence number is always 0, so it appears that Asterisk is not mis-displaying You shouldn't be *receiving* CNG, as you are the calling endpoint. If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS, etc. then something is badly wrong. ... and, that thing is probably the sequence number. Once Asterisk sees a packet with sequence number 0, any subsequent packets received with the same sequence number will be dropped (because according to the T.38 recommendation, they must be retransmissions... new packets would have higher sequence numbers). So these UDPTL packets are never making their way up to the FAX stack, and the FAX transaction never gets started. I guess it must be common for UDPTL stacks out there to just not care about repeated sequence numbers, although the one in Asterisk sure does (and it's based on the same code as the one in CallWeaver, FreeSwitch and maybe other packages too). If you'd like to experiment, you can comment out lines 495 and 511 of main/udptl.c, which will make Asterisk's UDPTL stack just not care at all about the incoming sequence number. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 04:42 PM, Kevin P. Fleming wrote: You shouldn't be *receiving* CNG, as you are the calling endpoint. You're right. Hadn't even thought about that. If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS, etc. then something is badly wrong. ... and, that thing is probably the sequence number. Once Asterisk sees a packet with sequence number 0, any subsequent packets received with the same sequence number will be dropped (because according to the T.38 recommendation, they must be retransmissions... new packets would have higher sequence numbers). So these UDPTL packets are never making their way up to the FAX stack, and the FAX transaction never gets started. I guess it must be common for UDPTL stacks out there to just not care about repeated sequence numbers, although the one in Asterisk sure does (and it's based on the same code as the one in CallWeaver, FreeSwitch and maybe other packages too). If you'd like to experiment, you can comment out lines 495 and 511 of main/udptl.c, which will make Asterisk's UDPTL stack just not care at all about the incoming sequence number. HEAD out of SVN + the sequence number change still gets no fax transmit. I do get a few addition fax status messages on the console, though. I'm getting -- FAX handle 0: [ 000.062327 ], STAT_EVT_RX_IMG_STRT st: WT_DIS rt: UNEXPECT 4-9 times (with changing timestamps, of course), then nothing until disconnect. It sounds like that Gafachi's T38 implementation is horribly, horribly broken. I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. -- _ -- 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
