[asterisk-users] AGI Perl Question
From inside a Perl script, Asterisk 1.4, I am trying to get this information
$xipaddress = $AGI-get_full_variable('CHANNEL(recvip)');
or using pan get_variable
But I get nothing. How do I read the IP address of origin from an AGI Perl
script?
I cannot update the version, for this is an old system that I am being paid
to keep it running.
Philip Orleans
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[asterisk-users] PSJIP Leak handle
I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also asterisk -rx core show channels, etc. I am returned to the bash prompt. I checked the handles and lsof | grep asterisk |wc -l 7098126 I think there is a kind of handle leak here. Nothing else runs in the box If there is a way to find out what happens, let me know. The dialplan is confidential, for it belongs to my customer,but I can give you access to the box. In short , the app receives a call, checks the number against a database and calls app_transfer. That is it. This is what I see when the command fails: asterisk -r Asterisk SVN-branch-13-r431555M, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = [root@centos7 /]# this command shows the issue, thousands of lines lsof | grep asterisk asterisk 4077 root *450w FIFO 0,8 0t0 110430221 pipe asterisk 4077 root *451r FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *452w FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *453r FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *454w FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *455r FIFO 0,8 0t0 110426507 pipe asterisk 4077 root *456w FIFO 0,8 0t0 110426507 pipe^ It looks like https://issues.asterisk.org/jira/browse/ASTERISK-823 but in fact I am using PJSIP. It is definitely PJSIP, for I replaced the dialplan with plain SIP, and there is no issue, ceteris paribus. Note: I am not a developer and have no idea how to troubleshoot this. -- _ -- 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] CALLERID(ani2) inserting
I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
this
sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
and it can be read by using
Set(var=${CALLERID(ani2)}
But how do we add that information to the outbound INVITE? This is
critical in the toll-free industry and call-from-jail industries.
Thanks for your help.
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[asterisk-users] New Feature CALLERID(ani2) read/write
Two years ago we added logic to parse the isup-oli parameters, that arrive
as part of the FROM Sip header. We need to finish the job and allow setting
of this parameter for outbound calling, both in traditional SIP channel and
PJSIP, which I believe will replace all instances of the old SIP channel
soon.
Right now, if we try to set CALLERID(ani2)=$
{var}
, there is a runtime error because this variable is read-only.
The business community around Asterisk needs this feature and there is no
known workaround.
I am also writing about this to the developer list. If somebody wants to
propose a patch, I can contribute to the bounty.
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[asterisk-users] Google Voice
Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -- _ -- 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] taskprocessor fails to allocate memory
I keep getting this error [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor bbe08c34-9d1c-4e5f-8ae0-0cc75289caca [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor f30fcb95-d290-4bb1-8008-290b79342c01 [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614 __allocate_taskprocessor: Unable to start taskprocessor listener for taskprocessor 38660bf7-eec2-4ce6-a9d7-63c8178a0556 [Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245 default_listener_shutdown: pthread_join(): Cannot allocate memory After 18 instances of Asterisk using parameter -C /etc/asterisk1/asterisk.conf -C /etc/asterisk2/asterisk.conf -C /etc/asterisk3/asterisk.conf etc. The machine has 180 GB of RAM and 16 cores. ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 1048576 max locked memory (kbytes, -l) unlimited max memory size (kbytes, -m) unlimited open files (-n) 1048576 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) unlimited virtual memory (kbytes, -v) unlimited resources is plenty free -h totalusedfree shared buff/cache available Mem: 177G 60G111G508K5.4G 116G Swap: 269G 0B269G and nothing else runs in the box I am using regular chan_sip Where do I go from here? Your help is appreciated. -- _ -- 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 12 is broken
The amount of threads went through the roof ls /proc/15373/task | wc -l 682 in version SVN-branch-12-r427618M it used to be 18 in Asterisk SVN-branch-11-r412226M How can I trace this? There are no calls open, on a disconnected system -- _ -- 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] One thread per peer
Is this normal to create one thread per peer in Asterisk 12, chan_sip regular, not pjsip? What happens is I have 659 peers, and I get 682 tasks on ls /proc/15373/task | wc -l If this is normal then of course I can only get a few instances before my box collapses. Is it any different in pjsip? -- _ -- 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] Ports leak
I am using Asterisk 12 svn, from today, and after a few thousand calls, I run out of ports. This happens eith PJSIOP and regular old SIP. I think it is RTP related. Any idea how can I troblshoot this. It happened teh same with Asterisk 11. On the other end there is a freeswitch. My guess is that there is an incompatibility. Thanks in advance for your thoughts -- _ -- 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] SIP 380 Alternative Service with PJSIP
I need to respond with 380 Alternative Service. Is there a way to do this in PJSIP? Please note that I am not picking up the call. For instance, the Transfer app closes the call if you did not answer it first. There is a bug open about this. I want to stay with PJSIP, for I found that it scales painlessly to 1000+ calls, basically, I have not found an upper limit yet. Thanks for your help. -- _ -- 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] PJSIP and Multiple transports per endpoint
I have a multihomed machine. How can I assign multiple IPs to and endpoint, not all of them, just two, for instance, out of many? Suppose the machine as 30 IPs, but my asterisk needs listen on two, and one single endpoint needs to be associated with those two IPs. I tried to add a second bind line to a transport, but it ignores all after the first one. I tried to add a second transport line to an endpoint, but it only considers one. Thanks for your help. -- _ -- 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] Channel h323 and oh323 fails to match inbound IP
I am having the issue described in this question: http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html Does anybody has an insight? I guess Asterisk is trying to match the combination IP:Port, but in H223 this changes call by call. There is no way to add insecure=port like in channel_sip. -- _ -- 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] PJSIP and Multiple transports per endpoint
I had some confusion here. The endpoint needs a transport in order to carry calls out. But the transports are also used by the application PJSIP at large, in order to listen for incoming connections. In order to just receive calls, I think you only need a transport, but no need to assign that transport to any endpoint. For example if you are just acting as voicemail or a pure IVR system. If you have a multi-homed machine, you need a transport for each IP where you expect to receive calls. Please correct me if I am 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
[asterisk-users] Question about SIP warning
I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream: audio 30660 RTP/AVP 4 101 13 What does it mean and does it show a problem like one-way audio? Thanks for your help. -- _ -- 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] Possible handle leak in PJSIP
I have been seeing errors saying the Asterisk cannot establish an RTP
connection, so I did this:
lsof -i -n -P | grep asterisk | wc -l
10483
but I have only
Asterisk 11 has 1 open calls
Asterisk 12 has 21 open calls
Asterisk 14 has 19 open calls
Asterisk 15 has 22 open calls
Asterisk 16 has 15 open calls
Asterisk 17 has 15 open calls
Asterisk 30 has 71 open calls
Total
164 active calls
The machine has 30 asterisk process, most of them dormant.
There is no way with 164 active calls we may have 10484 handles allocated.
I have no idea how to debug this. I suggest that an experienced
engineer from Digium logs into the box and researches this problem,
else nobody is going to ever be able to use PJSIP in production.
The box is Fedora 20, fully updated.
It grows about 30 handles per minute and it never goes down.
The dialplan is actually a four liner
look at the audiowritecodec
select an outbound endpoint based on that
The idea is to bridge calls based on the codec to avoid any
transcoding, so I have two outbound codecs and I dial like this:
exten = _X.,1,Set(_SIP_CODEC_OUTBOUND=${CHANNEL(audiowriteformat):0:4})
exten = _X.,n,Goto(${SIP_CODEC_OUTBOUND})
exten = _X.,n(ulaw),Dial(PJSIP/alawoutbound/sip:${EXTEN}@X.X.X.X)
exten = _X.,n(g729),Dial(PJSIP/g729outbound/sip:${EXTEN}@X.X.X.X)
As you can see, Houston, we have a problem
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[asterisk-users] Question about SIP Dial
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
but it does not work. I just cannot embed the IP address in the peer's
definition, but I need to use some other configuration features that
are unique to each peer.
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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
[asterisk-users] Asterisk 12 fails to launch with option -C
I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is the same as current 0 gl_pathc 0 == Sorcery registered wizard 'bucket' == Sorcery registered wizard 'bucket_file' Cannot update type 'bucket' in module 'core' because it has no existing documentation! Failed to register 'bucket' object type in Bucket sorcery The debug level is 5, and so is the verbose level. The logger.conf has this line myDebugLog = notice,warning,error,debug,verbose,dtmf And it does not even get created. /etc/asterisk1/asterisk.conf has these lines astetcdir = /etc/asterisk1 astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk1 astdbdir = /var/lib/asterisk1 astkeydir = /var/lib/asterisk1 astdatadir = /var/lib/asterisk1 astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk1 astrundir = /temp/run/asterisk1 astlogdir = /var/log/asterisk1 astsbindir = /usr/sbin All directories mentioned above do exist. Should I open a bug or there is something I am missing? -- _ -- 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] Native architecture never available in menuselect
I want to compile Asterisk always for the native architecture of the machine, and I find that it is never available. It says Depends on: native_arch(E) Can use: N/A Conflicts with: N/A Support Level: core This is Fedora 20 gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1) many thanks Philip -- _ -- 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] Need a developer to write me a patch
I cannot wait for the regular bug-patch process to play out. I am considering hiring a developer to fix bug 24015, and of course submit the patch for the bug. The issue is simple, the app Transfer does not transfer when using PJSIP.. I called Digium and they said that they do not do this kind of work. -- _ -- 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] PJSIP Transfer not working
I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer' -- Executing [17274428141@redirect:30] Transfer(PJSIP/Client.1.1.1.1-0002, PJSIP/17274428141;rn=+1813402;npdi@1.1.1.1) in new stack [Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869 pbx_extension_helper: Launching 'Verbose' -- Executing [17274428141@redirect:31] Verbose(PJSIP/Client.1.1.1.1-0002, 2,Transferred: 17274428141;rn=+1813402;npdi@1.1.1.1) in new stack == Transferred: 17274428141;rn=+1813402;npdi@1.1.1.1 -- Auto fallthrough, channel 'PJSIP/Client.1.1.1.1-0002' status is 'UNKNOWN' [Jul 9 21:39:29] DEBUG[47716][C-0002]: channel.c:2597 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'PJSIP/Client.1.1.1.1-0002' [Jul 9 21:39:29] DEBUG[47716][C-0002]: channel.c:2753 ast_hangup: Hanging up channel 'PJSIP/Client.1.1.1.1-0002' [Jul 9 21:39:29] DEBUG[47716][C-0002]: chan_pjsip.c:1578 hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP) --- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 --- SIP/2.0 603 Decline v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z- i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY f: 957408 sip:957408@8.26.191.189;tag=82c82c1d t: sip:17274428141@8.26.191.189;tag=09f3a67a-f457-46d1-8d16-243478ac3859 CSeq: 1 INVITE Reason: Q.850;cause=0 l: 0 -- _ -- 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] PJSIP incompatibility
Dear friends After spending few days converting my app to PJSIP, today I had to roll back the upgrade because in the SDP, the Owner section is wrong, or I misconfigured something This is what my client said: That OK message from 1.1.1.1 can not be parsed by our switch due to address representation in their SDP: Owner/Creator, Session Id (o): Pitcom 2723451647 2723451649 IN IP4 Pitcomlxc Such address representation not supported, there should be IP address instead of domain name. Example: Owner/Creator, Session Id (o): Pitcom 2723451647 2723451649 IN IP4 1.1.1.1 In fact, I traced it in the SDP packet, I see o=Pitcom 3991413436 3 IN IP4 pitcomlxc where pitcomlxc is the host name. How do I make PJSIP use an IP address there instead of the host name? My /etc/hosts.com has an entry for pitcomlxc, but it makes no difference. -- _ -- 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] PSIP Warning
Dear Friends I keep getting this warning [Jul 2 19:19:11] WARNING[16033][C-0441]: chan_pjsip.c:645 chan_pjsip_write: Can't send 10 type frames with PJSIP But I could not find an explanation by googling. Any idea? -- _ -- 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] PJSIP endpoint max-calls limit missing
I could not find a way to set a max on the calls allowed through a PJSIP endpoint. In case we decide to add it, the we need another reason for the call to fail in the Dial application, something like limit reached Am I missing this capability? -- _ -- 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] PJSIP question
In a PJSIP endpoint, how do I set all no-named settings so they get inherited from another place and I don't need to mention them again and again for all my endpoints? In regular sip you could specify those options and they remained valid if not redefined by a peer. A case would be the codecs allowed. I tried to include those global options in a section called [global] disallow=all allow=ulaw but the endpoints do not have knowledge of any such global options. -- _ -- 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] PJSIP Include not working
I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled? -- _ -- 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 12 and chan_local
I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened? -- _ -- 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] PJSIP Dial via IP fails
Dear friends
This is my simple dialplan
[demopjsip]
exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten = _X.,n,Hangup()
I need to dial out via an IP address, not using an endpoint, as shown above.
It fails with
Executing [1957408@demopjsip:3] Dial(PJSIP/federico-0002,
PJSIP/195XXX7408@10.10.10.2) in new stack
[Jun 26 00:39:00] ERROR[10136]: chan_pjsip.c:1722 request: Unable to
create PJSIP channel - endpoint '10.10.10.2' was not found
[Jun 26 00:39:00] WARNING[10167][C-0002]: app_dial.c:2421
dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No
route to destination)
I remember that this Dial format was possible with regular SIP. The IP
address is routable, so there is no specific network issue.
In my pjsip.coonf I defined a default outbound endpoint
[global]
default_outbound_endpoint=default_outbound_endpoint
In that default endpoint defined, I did not add any IP address,
because I want to keep it generic and dial any IP address with the
same settings,
Is this possible?
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[asterisk-users] PJSIP question
A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip -- _ -- 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] Inter-Digit delay when dialing out
Okay, I see that there is no way to shorten the inter-digit delay when we dial out DTMF. The reason I need this is because when dialing a two-stage call, the PDD can be shortened dramatically while still working fine, and that opens whole new world of business. But many people confused dialing-out with receiving DTMF, which is not what we are concerned with. If it is impossible to configure the time lapsed between digits in any configuration file (it should be done in asterisk.conf), the can somebody help me by pointing where in the source code can that be tailored? -- _ -- 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-users Digest, Vol 119, Issue 7
I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, asterisk-users-requ...@lists.digium.com wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Shorten time between DTMF (CDR) 2. Re: Shorten time between DTMF (Eric Wieling) -- Message: 1 Date: Fri, 6 Jun 2014 13:04:09 -0400 From: CDR vene...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Shorten time between DTMF Message-ID: CAC9cSOCzEWrsVXcq2=dffhjtc5ga6sgdujwigge84eu7rb1...@mail.gmail.com Content-Type: text/plain; charset=UTF-8 I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- Message: 2 Date: Fri, 6 Jun 2014 13:08:36 -0400 From: Eric Wieling ewiel...@nyigc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Shorten time between DTMF Message-ID: 616B4ECE1290D441AD56124FEBB03D082D165E9BEF@mailserver2007.nyigc.globe Content-Type: text/plain; charset=us-ascii Which EXACT parameter did you change in asterisk.conf? Changing DTMF duration for DAHDI is done in chan_dahdi.conf. SIP DTMF duration and inter-digit duration is generally set on the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR Sent: Friday, June 06, 2014 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Shorten time between DTMF I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- _ -- 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-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 119, Issue 7 ** -- _ -- 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] Shorten time between DTMF
I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- _ -- 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] Change time between DTMF
I already shortened the DTMF duration, but I need to change the time elapsing between them. The first thing I achieved by changing a parameter in asterisk.conf, but how do I conquer the second goal? -- _ -- 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] 500 Server Error on Null Caller ID
When a client send me an INVITE with this type of caller ID From: eurus sip:null@XX.XX.XX.XXX;tag=3430296121-3809549020-352327076-1077499159 Asterisk 14 sends back SIP/2.0 500 Server error occurred (1/SL) My client says Yes, I know the null is there but this not illegal and perfectly acceptable according to rfc 3261. Should I open a bug 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
[asterisk-users] Adding a SIP header to a reject 503
Is there a way to add an X-Header to hangup(34), which translates to a 503? I tried adding it before the hangup but it never gets transmitted -- _ -- 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] Help with a bug (CDR)
I fund the issue and it was in my own code. I apologize. -- _ -- 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] Help with a bug
Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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] Need to hire recordings for an IVR
I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer. -- _ -- 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] Why doesn't Asterisk try to prevent transcoding
I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes I vote for this new feature. However, I don't have the expertise to write a patch. I would say that only Digium developers could attempt to do this without disrupting the code too much. I also tried to migrate to PJSIP, but had to go back when I realized there was no channel variable contaning the inbound IP address. In general, any channel hast to provide the information to the dialplan, somehow, otherwise we cannot do business. I hope the PJSIP integration matures soon. -- _ -- 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 on Windows
Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. -- _ -- 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] File Leak Handle in 11.60
I am using Asterisk 11.6.0 built by root @ linux-t784 on a x86_64 The issue is a huge UDP handle leak, presumably coming fro ooh323 With 45 calls open calls (ooh323 to SIP), I have netstat -anp | grep asterisk | wc -l 6669 lsof -p 6785 -i -n -P | grep UDP | wc -l 6567 The machine needs to be rebooted as soon as Asterisk eats up all the handles. The question is, how can I further debug this? How do I know if this is in fact ooh323 or SIP or something else? Is ooh323 still supported? What kind of trace should I capture if I decide to file a bug? Would Valgrind help? Thanks for your help -- _ -- 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] Question about Management Interface
I am trying to identify the module (*.so) that contains the Asterisk Management Interface, so as to set noload=XXX.so in modules.conf. Any idea? -- _ -- 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] XMPP question
I need to log out from a Google Voice account, before I use a different account, otherwerwise Gvoice will block calling capability. How do I do that? I cannot figure this out. There should be function or application to cleanly log out Philip -- _ -- 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] Capture Media IP in CDR
Right now,there is no way know to capture the Media IP. The channel variable does not know about it. It requires adding anew variable to CHANNEL(), and also it entails to force every channel to update that variable. New channels like PJSIP do not even update the known variables like CHANNEL(recvip). So this is not trivial and in my opinion, only Digium may do this. Also, please remember that this IP can change dynamically along the way, via re-invites. More than once? Don't now the answer. If that is the case, then let's create a large variable and keep adding to it via concatenation and a field separator. I am not an officer, but I can see clearly when this patch will save maybe hundreds of lives. A crucial call will come and if we know where it came from, somebody is going to walk back home safe. -- _ -- 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] Capture Media IP in CDR (CDR)
I am quite surprised about the degree of surprise in the group. A few days ago, somebody called a school and issued a threat, through my network. The call came from China, but of course it was US caller. The DA wants to know where call came from. The caller ID is Restricted and the chinese carrier is playing games. If I had a way to store the media IP, I would be able to pinpoint the offender in the US, or the company that touched the media last. As a result of Asterisk not having this functionality, many children are danger and this country at large is at a great peril, since Asterisk is the most widely used low-cost technology for telecommunications. I need Digium to store this IP in the CDR. I will be honest with the government and let them know that my tool is incapable of saving lives or safeguarding our national security because nobody thought about this. PD: I am not paying for a patch, since this is huge burden on a small company like mine, with a single employee, and also because the whole world will enjoy the benefit. It is not fair that I would have to hire somebody to patch Asterisk. I appeal to Digium to patch Asterisk. -- _ -- 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] Capture Media IP in CDR
The CHANNEL() function has no idea about the media IP, and also SIP_HEADER(), since the media IP is not known until the call has been established and a reinvite has been received and dispatched. I am using of course, directmedia=yes and directrtpsetup=yes. Hence my question to the group. -- _ -- 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] Capture Media IP in CDR
I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay in business. -- _ -- 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] Function CDR_PROP
In Asterisk 12, how should I call the function CDR_PROP set(CDR_PROP(disable)=true) or simply CDR_PROP(disable) I am getting two records per call attempt, and I cannot figure out how to go back to get only one record. So far I am using this technique, but it changes nothing. My calls always involve a single caller a single calee exten = 100,1,NoOp() same = n,Dial(SIP/bob,,b(default^callee_handler^1)) same = n,Hangup() exten = callee_handler,1,NoOp() same =n,Set(CDR_PROP(disable)=true) same =n,Return() Philip -- _ -- 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] PJSIP Authrentication by IP fails
I have one single endpoint [inhouse](endpoint-basic) type=endpoint and one section like this [indentify] endpoint=inhouse match=X.Z.Y.X But when I call in from IP X.Z.Y.X, it does not match my peer. It tries to send the call the default context, and the number is not found. I know that I should have loaded this module, res_pjsip_endpoint_identifier_ip.so and it is loaded. In fact, all others authentitcation modules are not, since I only identify via IP. I did not load all the methods I will never use. load = res_pjsip.so load = chan_pjsip.so load = res_pjsip_acl.so noload = res_pjsip_authenticator_digest.so load = res_pjsip_caller_id.so load = res_pjsip_diversion.so load = res_pjsip_dtmf_info.so noload = res_pjsip_endpoint_identifier_anonymous.so load = res_pjsip_endpoint_identifier_ip.so noload = res_pjsip_endpoint_identifier_user.so load = res_pjsip_exten_state.so load = res_pjsip_log_forwarder.so load = res_pjsip_logger.so noload = res_pjsip_messaging.so noload = res_pjsip_mwi.so load = res_pjsip_nat.so load = res_pjsip_notify.so noload = res_pjsip_one_touch_record_info.so noload = res_pjsip_outbound_authenticator_digest.so noload = res_pjsip_outbound_registration.so load = res_pjsip_pidf.so load = res_pjsip_pubsub.so noload = res_pjsip_refer.so noload = res_pjsip_registrar_expire.so noload = res_pjsip_registrar.so load = res_pjsip_rfc3326.so load = res_pjsip_sdp_rtp.so load = res_pjsip_session.so noload = res_pjsip_t38.so noload = res_pjsip_transport_websocket.so load = res_pjsip_acl.so I need to identify the caller, and since there is no way to see the IP address of the caller in the dial plan, at least I need to force Asterisk to match it to a peer. Any ideas? -- _ -- 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] PJSIP Identify Wiky
The Wiky needs to be updated https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29 This is the example shown: [6001] endpoint=6001 match=203.0.113.1 It should be: [6001] type=identify endpoint=6001 match=203.0.113.1 -- _ -- 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] PJSIP question
I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type sip set debug on Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip -- _ -- 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] PJSIP question urgent
I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)},
so if I use PJSIP, for scalability, how do I read what the signalling
IP where the inbound call is coming from and what is the inbound
codec?
You would think that the new channel would set those variables up, isn't it?
Philip Orleans
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Re: [asterisk-users] asterisk-users Digest, Vol 109, Issue 30
I am stumped
In features.conf,I programmed this
[applicationmap]
Answer0 = 0,self/both,Macro,nway_start
But do I pass an argument or parameter to my macro? I tried
Answer0 = 0,self/both,Macro,nway_start^0
Answer0 = 0,self/both,Macro,nway_start,0
but the usuar variable ${ARG1} is empty in my dialplan.
The issue is that my macro needs to know what key was pressed.
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[asterisk-users] Question about media before connect
I need to block any audio before there is a connect, in SIP. How do I tell the DIAL application to behave like that? Yours Philip -- _ -- 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] Question
Is it me or Google just blocked Asterisk's chan_motif? I get violation of terms of service audio message whenever I send a call. Philip -- _ -- 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] Loopback question
Dear friends I need to loopback the audio on my channel. Did anybody on the development team thought about a function or app that would do that? If it is not clear, I mean that whatever audio I get, I send back. Philip -- _ -- 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] Is it a BUG
In safe_asterisk, there is a section where the script executes some startup scripts, located in /etc/asterisk/startup.d However, when you restart asterisk with core restart now or you go ahead and kill the asterisk process, these scripts that are so important do not get executed. The question is: where in the safe_asterisk script can I copy the whole loop so in any event, if Asterisk gets restarted, these scripts get properly executed. Otherwise there is no way to ensure that the finite-state machine starts from an known start point. -- _ -- 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] Bug or Not
I think I found another bug, but please let me know if there is a workaround, since my bugs never get fixed. In safe_asterisk, there is a section where the script executes some startup scripts, located in /etc/asterisk/startup.d However, when you restart asterisk with core restart now or you go ahead and kill the asterisk process, these scripts that are so important do not get executed. The question is: where in the safe_asterisk script can I copy the whole loop so in any event, if Asterisk gets restarted, these scripts get properly executed. Otherwise there is no way to ensure that the finite-state machine starts from an known start point. -- _ -- 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] Please dont tell me this is impossible
I have been fighting all night with version 1.8 and have not found a way to do this with any command or Perl AGI-command. I need to play a file and wait until the customer presses at least $maxdigits to return, BUT, the file must continue playing until $maxdigits is received or $timeout has expired. So far I found impossible to achieve this functionality. Am I missing something? Philip -- _ -- 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] Please dont tell me this is impossible
This is from the documentation of Perl-AGI
$AGI-stream_file($filename, $digits, $offset)
Executes AGI Command STREAM FILE $filename $digits [$offset]
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI-stream_file('demo-echotest', '0123');
$AGI-stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed
It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip
On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip
The Playcommand will be interrupted by the key but the agi result contains
the offset. So you can play this file from offset again until you $maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE
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[asterisk-users] SSM
I need to send SMS from Asterisk to an SMPP server. Is there a SMPP channel or any other know way to send SMS via Asterisk? I don't care if its is paid software. Philip -- _ -- 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] Is there any way to make call fail after # of rings?
My customer needs to set a forwarding based on number of rings,i.e., if the phone rings 5 times (user-selectable), then try another number. Is there a way to do such a thing with Asterisk? I could not find way to do it based on the documentation of the Dial function. The protocol is SIP only, however, I could use a different one if it provided a workaround. If this is the wrong tool for the job, what technology would do this? -- _ -- 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] Stumped about adding a semi-colon to a variable
I need to add a semi-colon to a variable, but no matter how I quote it, the parser ignores it and considers the semi-colon as the beginning of a comment. Si how do I concatenate the content of a variable to a semi-colon? I tried surrounding it with double quotes, single quotes, using a backslash first, a period first, to no avail. -- _ -- 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] Question about Registrations
In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? Venefax -- _ -- 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] Message prints even if verbose level is Zero
In 1.8, somebody left a message that shows up like this Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457 It could be also Local Bridging The point is that this message should not print in the console unless the verbose level reaches some level. Never at level zero. It should be a notice, etc. When there is a lot of traffic, this message consumes CPU unnecessarily. -- _ -- 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] Timer B in sip.conf cannot be changed
I am using 1.8. I need to change timerb to 6500, that is, if there is no response of some sort in 6.5 seconds, consider the call failed and try another route. It does not matter what do I set for the other timers: T1min=100 timert1=100 Timerb=6500 The command sip show settings always shows Timer B=32000. Any ideas how can I reduce Timer B? -- _ -- 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] Version 1.8 strange expression error
This expression that worked fine in 1.6.2 is returning an error:
exten =_X.,n,Set(i=$[${i} + 1])
It needs to add 1 to the value if i. Did I miss something?
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[asterisk-users] 1.8 issues with Local Bridging
I encourage the developers to check this out http://forums.asterisk.org/viewtopic.php?f=1t=77692p=161590#p161590 I am calling from behind a NAT, and there is no way to force Asterisk to stay in the path. If the codec is the same as the outbound leg, it always does Remote bridging, but of course, creates a 1 way audio. I tried everything in the book directrtpsetup=no directmedia=nonat canreinvite=nonat and directrtpsetup=no directmedia=no canreinvite=no But it just behaves different than in 1.6.2 Any ideas how to make sure that the NAT works? -- _ -- 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] Securing Asterisk
This is turning into a political issue such as the one in Washington and the impending default on US debt. The point is that a minor change in the code would have a dramatic effect on security, and carry a lower impact on CPU that using Iptables. The simplicity of the change cannot understated. The hackers do not continue sending packets with new REGISTER attempts unless they see a response. The would move on. Digium is being monarchical about this. It looks like a loss of contact with reality. The vast ecosystem of Digium is made of hundreds of people like me. I am being forced now to place Opensips in front of Asterisk, in port 5060, set Asterisk to listen at Port 5061, and block access to 5061 from outside. Instead of a minor change, I have to bring a second application to the picture. The reason why I find useless using iptables and a rule that bans an IP address if it communicates more than a threshold of times, is simple. I have customers that hit me 10+ times per seconds from the same IP. It would look like a hacker, and it is not. I use a cluster of Asterisk in the same box, a big server, and each asterisks listens in its own network interface, and responds from it. It does work. But iptables or fail2ban would not work in a wholesale scenario. Any way, thanks for your attention. -- _ -- 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] Securing Asterisk
Only way to cope with hackers would be that Digium comes to its senses and accepts to disable any response to a REGISTER whose username is unknown. I cannot think of a good reason why Digium finds this proposal unacceptable, given the onslaught of hacking that we are seeing in the industry. It may take a single line of code and it would save millions of $$$. Not only because the hackers will never get in, but because we would save a huge CPU impact responding to hundreds of REGISTER attempts per minute. It is a NO brainer. Can please the Powers that Be reconsider and add this option to sip.conf? Please? -- _ -- 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] Securing Asterisk
I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico On Sat, Jul 23, 2011 at 1:00 PM, asterisk-users-requ...@lists.digium.com wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: use dahdi for local terminal modem access? (Lyle Giese) 2. dialplan pattern help (Armand Fumal) 3. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Patrick Lists) 4. Re: Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined (Paul Belanger) -- Message: 1 Date: Sat, 23 Jul 2011 09:29:26 -0500 From: Lyle Giese l...@lcrcomputer.net Subject: Re: [asterisk-users] use dahdi for local terminal modem access? To: asterisk-users@lists.digium.com Message-ID: 4e2adac6.4010...@lcrcomputer.net Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 07/22/11 22:47, William Stillwell wrote: Um, no VOIP involved here. Wrong. What do you think Asterisk is? Chopped meat? It's a VoIP switch. All traffic inside Asterisk is VoIP. I have an asterisk server with 2 23B+D PRI's I want to telnet/ssh into the asterisk server, and make an outbound call serial based modem/terminal connection (Like the 80/90's BBS Days). No TCP/IP or PPP or crazyness (ie, dialing into a Modem set to AA hooked to a Cisco Console Port) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Friday, July 22, 2011 8:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use dahdi for local terminal modem access? On 07/22/11 18:13, William Stillwell wrote: I have some terminals that have phone lines. One of my tech had an idea of using IAXmodem or something similar to use existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console. Anybody ever heard of doing this? I would think maybe would use iaxmodem maybe and a shell terminal app? (basically I'm dialing into a remote access device that uses a pots like for remote administration, and don't want to string a channel bank off my asterisk box, and a hook to a modem) -- Depends on your expectation. Because of compression in the codecs, it will be hard to get fast dialup. If you mean ssh or telnet, it might work. If you mean vnc or RDP over this, you may not get enough usable bandwidth to do that. Given this, I have in an emergency dialed into a RAS server via a VoIP line. My laptop connected at 14,400bps. All I needed to do was telnet into an APC masterswitch to toggle power on one outlet. It worked. I was surprised at getting a 14,400bps connect. I was not expecting that high and really did not need that high. 300 baud probably would have been fast enough to telnet into an APC masterswitch. Lyle Giese LCR Computer Services, Inc. -- _ -- 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 -- Message: 2 Date: Sat, 23 Jul 2011 14:30:42 + From: Armand Fumal a...@cybernet.lu Subject: [asterisk-users] dialplan pattern help To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
[asterisk-users] Using Firewall to protect Asterisk
I need to keep out all connection from 5 countries, which originate most of the Denial of Service attacks. The entries are around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter way to do this by using User Tables in iptables, that will keep the speed equal to LOG(x). I already tried using a straight list and it kills the box. Unless a smarter way us found, there is no way to use iptables. Federico -- _ -- 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] Senddtmf inside a macro
Is it my imagination of the application senddtmf does not work inside a macro? Should I open a bug case or this is by design, and if so, what are the grounds for that decision? I called myself and no matter what I do, I cannot hear the tones, but if I place them inside the D(XXX) option of the dial command, I hear them. This issue makes impossible a set of application where complex negotiations between systems take place via DTMF. Federico -- _ -- 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] Supported: ms-early-media
Is there an option in sip.conf for 1.6.2 that would add this to the INVITE? Supported: ms-early-media In my invites I see: Supported: replaces, timer But I have not seen any option that would add the ms-early-media option. Here’s a link to the RFC3960 Describing the benefits of early media vs late media.In summary you eliminate “clipping” as it’s called. Media clipping occurs when the user (or the machine generating media) believes that the media session is already established but the establishment process has not finished yet. The user starts speaking (i.e., generating media) and the first few syllables or even the first few words are lost. -- _ -- 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] Unexplained message in 1.6.2
Every time I start Asterisk or do a simple reload I see this message: Cannot open maximum file descriptor 32767 at boot? No such file or directory Does anybody have some idea of what can it be? It did not happen in version 1.4. Philip -- _ -- 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] Unexplained message in 1.6.2
Every time I start Asterisk or do a simple reload I see this message: “Cannot open maximum file descriptor 32767 at boot? No such file or directory”. It only works if I set 1024 in asterisk.conf maxfiles However, my sysctl fs.file-max fs.file-max = 65535 and my ulimits are ulimit -a core file size (blocks, -c) unlimited data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 40 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 40 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 1056768 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited -- _ -- 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] How do I access the Dialstatus numeric code received?
Tilghman Lesher wrote
Not available in anything other than trunk (to be 1.8). It depends upon a
new
feature, so it's not something you can easily backport. After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
In a real dialplan, how do I get a variable with channel-name? I mean: My
app is Sip2Sip. I get one call inbound, get the number dialed ${EXTEN} and
proceed to try many carriers. If the carriers send me something different
than 503 Service Unavailable or 404 Not Found, I need to close the call and
send back whatever SIP code I got, exactly. There is no way for me to do
that now. Unless I am missing something, I can only play with ${DIALSTATUS}
and do Hangup(Code), but my Code variable is never the same that I got
from the second leg. I would like to be able to do
Hangup( ${HASH(SIP_CAUSE,channel-name)}, where channel-name is the last
channel used to dial-out. How do I do this in trunk? I will have to start
using trunk in production. Another issues is the the function Hangup(Code)
takes a decimal, not related to the SIP code I just got. How would you
design your 1.8 or 1.62 dialplan around this issue?
Thanks in advance.
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[asterisk-users] How do I access the Dialstatus numeric code received?
I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -- _ -- 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] How to use one single IP as origination
I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? -- _ -- 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] Calls per second limit in manager
My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- _ -- 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] Sending back the BYE code gotten on second leg
I have a business problem that is killing me. I do SIP2SIP, only. I place a call after receiving the incoming request, and I need to send a Hangup(Code) to the caller, based on the result of the outbound leg. How can I do that in Asterisk? Is that even possible at all? I can use Hangup(code), but how do I extract it from the received BYE? This is only for calls that fail to connect on the outbound leg. Philip -- _ -- 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
