Re: [asterisk-users] incoming calls fall into echo test mode
On Saturday 19 Jul 2014, Norman Molhant wrote: I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? You really have supplied incomplete information here, by neglecting to mention the actual extension number which is causing the problems. That would have had somebody onto it like a shot. What follows is an educated guess based on the most likely scenario according to the available information: Somewhere in your dialplan, probably in a section that has already been helpfully configured for you by FreePBX, the extension number you assigned to your customer has been appropriated for an echotest. I suggest to grep for (firstly) the extension number in question, and (if that does not work, perhaps because the echotest is a wildcard match aot a literal one) then search instead for 'exten[ ]*=' (afraid that one will give you many more hits . you'll have to look through them yourself) under /etc/asterisk. Use the -R option to search subfolders as well. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] chan_motif / res_xmpp problems
I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the -4 command line option to use IPv4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming session-initiate XML stanza but Asterisk does not send any XML back. I definitely have icesupport=yes in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody? -- _ -- 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] TLS, STRP and ARA
Hi I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is completely absent and tls is not an option for transport. Does this mean I can't configure a peer to use TLS and SRTP if using ARA? Are there any workarounds? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 14.4.0 MeetMe crash
Hi, after update on 12.4.0 asterisk crashes on MeetMe ending on 12.3.2 it worked well. Is some one else have this issues? should someone open a ticket? -- _ -- 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] Asterisk 14.4.0 MeetMe crash
On Mon, Jul 21, 2014 at 5:53 AM, Nick Awesome jl...@me.com wrote: Hi, after update on 12.4.0 asterisk crashes on MeetMe ending on 12.3.2 it worked well. Is some one else have this issues? should someone open a ticket? 1. There were no changes to MeetMe in 12.4.0: http://svn.asterisk.org/svn/asterisk/tags/12.4.0/asterisk-12.4.0-summary.txt 2. That being said, crashes are obviously bugs. Please generate a proper backtrace using the instructions on the wiki [1] and open an issue on the issue tracker [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org 3. Finally, you should know that MeetMe is an extended support module in Asterisk 12 [3]. Development support for it comes from the community. Depending on the nature of the crash, response times may reflect that. You may want to consider switching to ConfBridge, which is a core supported module in Asterisk 12. [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] chan_motif / res_xmpp problems
Daniel Pocock wrote: I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the -4 command line option to use IPv4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming session-initiate XML stanza but Asterisk does not send any XML back. I definitely have icesupport=yes in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody? What does your motif.conf configuration file contain? If it is not configured then it will not be associated with the account and the Jingle support will not be present. -- Joshua Colp Digium, Inc. | 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 -- 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] Hold ,UnHold Via AMI
Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -- _ -- 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] chan_motif / res_xmpp problems
On 21/07/14 14:33, Joshua Colp wrote: Daniel Pocock wrote: I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the -4 command line option to use IPv4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming session-initiate XML stanza but Asterisk does not send any XML back. I definitely have icesupport=yes in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody? What does your motif.conf configuration file contain? If it is not configured then it will not be associated with the account and the Jingle support will not be present. It is largely based on the default config: [default](!) disallow=all allow=ulaw allow=h264 context=incoming-motif ; Default context that incoming sessions will land in ;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer ;maxpayloads = 30 ; Maximum number of payloads we will offer [asterisk](default) disallow=all allow=alaw allow=ulaw transport=ice-udp connection=asterisk context=incoming_xmpp and in xmpp.conf: [asterisk] type=client serverhost=some-host username=asterisk@some-host secret=-- usetls=yes usesasl=yes status=available statusmessage=I may be available timeout=5 -- _ -- 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] Hold ,UnHold Via AMI
Probably you should use “Action: Park example: Action: Park Channel: SIP/1000-0003 Channel2: SIP/1000-0004 On 21 Jul 2014, at 17:00, mahdieh saeed mahdieh.sa...@gmail.com wrote: Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -- _ -- 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] TLS, STRP and ARA
I have just answered my own questions and it's all fine.
transport will accept a value of tls and interpret it (you'll have to alter
the column definition if you're using an enum).
encryption column can be added and interpreted, here's the column defintion
I used.
alter table sip add column encryption enum ('yes','no') default 'no';
On 21 July 2014 11:31, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP.
However, we exclusively use the asterisk realtime architecture using the
mysql connector.
Looking at tutorials we have to set encryption=yes and transport=tls for
any peer we want encrypted traffic for.
Having a look at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is completely absent and tls is not an
option for transport.
Does this mean I can't configure a peer to use TLS and SRTP if using ARA?
Are there any workarounds?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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[asterisk-users] Certified Asterisk 11.6 Menuselect
Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks, Ryan -- _ -- 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] Hold ,UnHold Via AMI
Thanks for your answer. It works. On Mon, Jul 21, 2014 at 5:56 PM, Nick Awesome jl...@me.com wrote: Probably you should use “Action: Park example: Action: Park Channel: SIP/1000-0003 Channel2: SIP/1000-0004 On 21 Jul 2014, at 17:00, mahdieh saeed mahdieh.sa...@gmail.com wrote: Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -- _ -- 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
[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.9.2 Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.2 DAHDI-Tools-v2.9.2 dahdi-linux-complete-2.9.2+2.9.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete * Mostly stability patches affecting the xpp, wctc4xxp, and wcte43xp drivers. * Allows compilation against CentOS 5.5 * Includes fixes for a crash when setting a tone zone on a channel currently playing a tone and spurious signaling bit transitions in wcte13xp and wcte43x drivers when running dahdi_cfg repeatedly. Shortlog of dahdi-linux changes since v2.9.1: Doug Bailey (1): wct4xxp: AMI w/CAS errata applies to octal card as well. Oron Peled (10): xpp: fix failed multi-PRI E1-T1 transition xpp: new xbus attribute: dahdi_registration xpp: prevent double dahdi un-registration xpp: stability fixes - xusb mutex xpp: stability -- cleaner xpp_open/close xpp: stability -- better debug information xpp: stability -- deadlock in waitfor_xpds() xpp: stability -- better xbus shut down xpp: demote some NOTICE() to DBG() xpp: re-organize calls so worker_reset() Shaun Ruffell (58): wcte13xp: Trivial. Remove duplicate pointer to struct pci_dev. wcte13xp: Remove redundant call to synchronize_irq(). wcte43xp: Close potential unbalanced call to enable_irq(). dahdi: Define pf_fmt() globally in kernel.h wctc4xxp: Trivial. Remove unused timer_list from struct tcb. wcte43x: Trivial fix of 'source' in comment. wcte43x: Build against 2.6.18 and CentOS 5.5 wctc4xxp: Make sure we call the pci_enable_mwi() function. wctc4xxp: Disable read-line and read-line-multiple PCI commands. wctc4xxp: Halt the card when an alert is received. wctc4xxp: Remove unused debug ioctl interface. wctc4xxp: Replace channel semaphore with channel mutex. wctc4xxp: Enable the fatal bus error interrupt. wctc4xxp: Always ack a response packet. wctc4xxp: Check for shutdown after acquiring the mutex lock. wctc4xxp: Do not need locks on the transcoder buffers. wctc4xxp: Do not allow duplicated sequence numbers to be received for the channels. wctc4xxp: Only capture commands once they are on the descriptor ring. wctc4xxp: We always want to ack the responses. wctc4xxp: Encode the function in the ACK. wctc4xxp: All the commands do not need to have completions embedded in them. wctc4xxp: Cleanup RTP for unopened channels. wctc4xxp: Trivial removal of the receiveprep function. wctc4xxp: Reduce the number of locks grabbed when sending commands wctc4xxp: Make sure csm_encaps commands are sent before RTP. wctc4xxp: Use hardware timer for polling and not kernel timer wctc4xxp: channel count does not need to be atomic. wctc4xxp: Allow the tx and rx descriptor rings to be different sizes wctc4xxp: Add debug option to print channel stats to kernel log. Add #include linux/slab.h to all files that call kzalloc|kmalloc|kfree. pciradio: interruptible_sleep_on_timeout() - msleep_interruptible() wctc4xxp: Speed up channel setup / tear-down. wctc4xxp: Handle all known interrupts regardless of mask. wctc4xxp: Speed up the rate of polling. wctc4xxp: Fix the timestamp calculation for the RTP stream. wctc4xxp: Service tx ring in interrupt handler. wctc4xxp: Prevent exhausting memory in firmware. wctc4xxp: Trivial fix typo that was preventing firmware load. wctc4xxp: Reload the firmware if a fatal alert was received. wctc4xxp: Constrain RTP payload to 500 bytes. wctc4xxp: Trivial removal of unused structure members. wctc4xxp: Trivial reduction of indentation level in wctc4xxp_watchdog() wctc4xxp: spin_lock() - spin_lock_irqsave() in wctc4xxp_watchdog() dahdi: Protect echocan creation/destruction with mutex. dahdi: dahdi_chan.ec_factory can be protected with the mutex. wct4xxp: Move bottom half processing from tasklet to workqueue. oct612x: Implement the SerializationObject callbacks. wct4xxp: Trivial kmalloc + memset - kzalloc. wct4xxp: Remove unused open/close span_ops callbacks. tor2: Remove unused open/close callbacks. Do not call dahdi_span_ops.open with spinlock held. wcte13xp: Do not get stuck in Not Open state when DAHDI_CONFIG_NOTOPEN is set. wcte43x: Change span flags to atomic bitfield. wcte43x: Do not get stuck in Not Open state when DAHDI_CONFIG_NOTOPEN is set. wct4xxp: Report rx signalling bit changes after spanconfig. dahdi: Stop tones on channel when updating tone zone. wcte13xp: Do not reconfigure framer when span lineconfig is not changed. wcte43x: Do not reconfigure framer when span lineconfig
[asterisk-users] Call Identifier Logging
Hello,
I am working on upgrading from Asterisk 1.8 to Asterisk 11.6. One of the
features we are excited for is Call Identifier
Logginghttps://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging.
However, it doesn't appear that this new Call ID is accessible from the dial
plan. Ideally we would like to store this Call ID in the CDR. Does anyone know
if this is possible?
I could do something like this, but it seems like a terrible hack:
same = n,Set(CALLID=${SHELL(asterisk -rx core show channel ${CHANNEL} | grep
' Call Identifer' | egrep -o 'C-[0-9a-f]+')})
Also as a side note, in the core show channel output ' Identifier' is misspelt
as ' Identifer'
Steven Wheeler
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[asterisk-users] Question about PJSIP
I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command pjsip reload was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong? -- _ -- 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] Question about PJSIP
On Mon, Jul 21, 2014 at 7:00 PM, CDR vene...@gmail.com wrote: I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command pjsip reload was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong? We routinely run multiple Asterisk instances on a single machine using the PJSIP stack. A log showing messages why the res_pjsip_* modules couldn't be loaded on a particular instance of Asterisk would be helpful. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
