[asterisk-users] local channels and g729a voice quality
Hi, We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (caller's channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on low volume. If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder. I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. -- _ -- 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] ssh to a Cisco 7961 is not working
Ken, Thank you for posting the details. The method worked perfectly. I was about to give up on connecting via SSH to manually provisioned Cisco phones. Thank you, Vladimir On 1/15/2012 8:52 PM, Ken Alker wrote: Flavio, Thank you for pointing this out! I was using the reference Configuring Settings on the Cisco Unified IP Phone and it spoke nothing of SSH so I ASSuMEd that meant Cisco wasn't acknowledging the ability for the phone to do SSH. Your research set me on the proper path. It turns out there are now (with current firmware) a couple of variables that must be added to the XML file. For anyone else struggling with this problem, here are two links referencing the necessary modification (the second is not in English but is the only example of a complete XML file that I found): http://stackoverflow.com/questions/7148543/cisco-7945-sip-and-sip-notify-problem http://arbeitsplatzvernichtung-durch-outsourcing.de/marty44/fritzcisco7970.html The bottom line is that I had to add the following to the vendorConfig section (and reboot a couple of times): sshAccess0/sshAccess sshPort22/sshPort Thanks again, Ken Impulse Internet Services http://www.impulse.net --On January 15, 2012 8:03:30 PM -0200 Flavio Miranda flaviormira...@hotmail.com wrote: Ken, According with cisco docs, ssh is disable by default: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/firmware/9_2_1/engli sh/release/notes/7900_921.html SSH Access The SSH Access settings option allows the administrator to enable or disable the SSH port on the phone using Cisco Unified CM Administration. When enabled, it allows the phone to accept the SSH connections. Disabling the SSH server functionality of the phone blocks the SSH access to the phone. This setting is disabled by default. This feature is supported on the following Cisco Unified IP Phones (SCCP and SIP): • [Image: height=] Cisco Unified IP Phone 7906G • [Image: height=] Cisco Unified IP Phone 7911G • [Image: height=] Cisco Unified IP Phone 7931G • [Image: height=] Cisco Unified IP Phone 7941G • [Image: height=] Cisco Unified IP Phone 7941G-GE • [Image: height=] Cisco Unified IP Phone 7942G • [Image: height=] Cisco Unified IP Phone 7945G • [Image: height=] Cisco Unified IP Phone 7961G • [Image: height=] Cisco Unified IP Phone 7961G-GE • [Image: height=] Cisco Unified IP Phone 7962G • [Image: height=] Cisco Unified IP Phone 7965G • [Image: height=] Cisco Unified IP Phone 7970G • [Image: height=] Cisco Unified IP Phone 7971G • [Image: height=] Cisco Unified IP Phone 7975G Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 13:32:36 -0800 From: k...@impulse.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ssh to a Cisco 7961 is not working Flavio, Thank you for your response. According to various wiki's (voip-info.org included), the 7961 is supposed to accept SSH connections (and in fact, many people recommend this for debugging, but what I often see is just connect via SSH as if it should simply work; I haven't run across any data indicating people have had problems connecting via ssh as I am). I must assume that either the wiki's are wrong (doubtful, but possible), or Cisco deactivated ssh in this firmware build, or I need to alter a setting in my SEP*.cnf.xml file or on the phone itself; but I don't know what that would be. As per below, I've defined an ssh userid and password via the xml file. --On January 15, 2012 10:20:06 AM -0200 Flavio Miranda flaviormira...@hotmail.com wrote: Ken, Does your phone is realy able to accept ssh connection? I mean , it is set up for it ? As we can see in the log, it is sending reset to the ssh client. 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 look like it is not accepting ssh connections. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 01:06:34 -0800 From: k...@impulse.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] ssh to a Cisco 7961 is not working I am trying to ssh to my Cisco 7961 VoIP phone (computer and phone on the same LAN and switch) but I always get a connection refused. I have tried from my desktop and a laptop running different OS's. I have tried ssh 10.0.0.155 and ssh cisco@10.0.0.155 from a command prompt. Here are the results from sniffing via Wireshark: 11038 2272.240571 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11039 2272.240681 10.0.0.172 10.0.0.155 TCP 78 57665 ssh [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=8 TSval=963558895 TSecr=0 SACK_PERM=1 11046 2272.241550 10.0.0.155 10.0.0.172 TCP 60 ssh 57665 [RST, ACK] Seq=1 Ack=1 Win=8192 Len=0 11047 2272.241554 10.0.0.155 10.0.0.172 TCP 60 ssh 57665
[asterisk-users] echo audio delay in SIP VOIP
Hello sir, There is an echo problem in sip voip call. I think it is because of delay in audio. Let me try to explain you my system setup. I have test asterisk on two different system. System : 1 OS :Ubuntu(10.04)Lucid System Type :x64-based pc Processor :Intel(R) Core i5 CPU M520 @ 2.40GHz RAM :4.00 GB System : 2 OS :CentOS (5.7) System Type :x86-based pc Processor :Intel(R) Pentium D CPU 3.00GHz RAM :1.00 GB I use beetel magiq android tablet as video sip dialer.( http://beetel.quasar.in/magiqII ), which is rebranded version of huawei ideos S7 tablet (http://www.huaweidevice.com/resource/mini/201008174756/ideos/products_s7.ht ml) I use D-link DIR-615 WIRELESS N 300 ROUTER (http://www.dlink.co.in/products/?pid=349) as wireless access point. I have try two android video softphone on my tablet. 1 Voipswitch : http://voipswitch.com/en/products/softphones/mobile-softphones/softphone-for -android/ 2 Linphone (1.2.2) : http://www.linphone.org/eng/download/packages/android.html * i also try both codec( G729 , G711) * android tablets and asterisk server are connected in Local Area Network(LAN). -- _ -- 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 as UAC: How to put call OnHold
Hi! Many thanks for this hint. I will try this! :-) A quick question: when doing this with MusicOnHold(): will the SIP server be aware that the call is placed onHold (i.e. will Asterisk send the mentioned re-INVITE)? The point is - if possible - we want the caller to hear the OnHold Music from the SIP server. If not we would have to copy the MoH to our Asterisk (and change it on our side too, when it changes at the SIP-server). Kind regards, John 2012/1/16 Sammy Govind govoi...@gmail.com Hi, yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed. Regards, Sammy On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng john999...@zweng.at wrote: Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: -- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute a=sendonly or a=inactive added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) (http://tools.ietf.org/html/rfc5359#section-2.1) where Bob would be my Asterisk (as UAC), Alice is the external caller and Proxy is the provider's SIP server. Question: -- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like Hold? Kind regards and greetings from Austria, John :-) -- _ -- 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 check currently used libraries from command line ?
Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to check currently used libraries from command line ?
ldd 2012/1/16 Olivier oza_4...@yahoo.fr Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 to check currently used libraries from command line ?
Hi Olivier, I suppose you give strace a try. It's a powerful debugging utility, you should be able to find everything you are looking for. best regards, Ruben Am 16.01.2012 11:14, schrieb Olivier: Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Server-to-server BLF
Hi to all, I've managed to get the XMPP PubSub method to work on my set-up! Just carefully follow these instructions on the wiki: https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub Maybe this IRC log would also help you troubleshoot: http://apt.rikers.org/%23asterisk-bugs/20091008.html.gz One thing I noticed though is that if you do a devstate list, the state is sometimes not the same as listed in core show hints (core show hints has the correct state). Nevertheless, BLF works good for me. BTW, has anyone on the list tried out the AIS method yet? I'm a bit curious which method is better. Regards, Ronald On Fri, Jan 13, 2012 at 3:44 PM, Leandro Dardini ldard...@gmail.com wrote: Me too, an maybe other people on the list are interested in knowing your effort result and maybe appreciate a guide on the topic. Thank you Leandro 2012/1/13 Ronald Cepres rbcep...@gmail.com: Hi Ishfaq, Thanks for your reply. I've already started trying the XMPP method so I can't help you with the AIS method as of the moment. I'll let you know the result of my test. Regards, Ronald On Fri, Jan 6, 2012 at 5:14 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Ronald I took a bit of interest in your problem as I'm going to have to be doing the same thing in a few weeks. oenais is in the yum repositories so you can install from there if using redhat/centos based OS It is also in apt repositories if you're using a debian based OS Let me know how you get on Ish On Thu, 2012-01-05 at 12:07 +0800, Ronald Cepres wrote: Hi Kevin, Thanks for your suggestion. On the website of OpenAIS, it seems that it is not supported anymore and their download links (FTP and SVN) are broken (been trying it for about a month now). Is it still possible to use OpenAIS method? The other solution on the wiki is using XMPP which is for jabber. IMHO, it means that the XMPP solution can't be used on SIP peers, right? Regards, Ronald On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 11/16/2011 04:18 AM, Ronald Cepres wrote: Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Here is one way: https://wiki.asterisk.org/wiki/display/AST/Distributed+Device +State+with+AIS There are other methods documented on the wiki as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Asterisk as UAC: How to put call OnHold
Hey, I have never worried about looking at the SIP re-invites or anything when we engage MoH() application in asterisk. You can do a quick test on your test machine for this. Regards, Sammy On Mon, Jan 16, 2012 at 2:57 PM, Johannes Zweng john999...@zweng.at wrote: Hi! Many thanks for this hint. I will try this! :-) A quick question: when doing this with MusicOnHold(): will the SIP server be aware that the call is placed onHold (i.e. will Asterisk send the mentioned re-INVITE)? The point is - if possible - we want the caller to hear the OnHold Music from the SIP server. If not we would have to copy the MoH to our Asterisk (and change it on our side too, when it changes at the SIP-server). Kind regards, John 2012/1/16 Sammy Govind govoi...@gmail.com Hi, yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed. Regards, Sammy On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng john999...@zweng.at wrote: Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: -- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute a=sendonly or a=inactive added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) ( http://tools.ietf.org/html/rfc5359#section-2.1) where Bob would be my Asterisk (as UAC), Alice is the external caller and Proxy is the provider's SIP server. Question: -- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like Hold? Kind regards and greetings from Austria, John :-) -- _ -- 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] How to check currently used libraries from command line ?
2012/1/16, Anton Kvashenkin anton.juga...@gmail.com: ldd Thanks for replying. I got this: # ldd /usr/sbin/asterisk linux-gate.so.1 = (0xb7886000) libssl.so.0.9.8 = /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7834000) libcrypto.so.0.9.8 = /usr/lib/i686/cmov/libcrypto.so.0.9.8 (0xb76dc000) libc.so.6 = /lib/i686/cmov/libc.so.6 (0xb7595000) libxml2.so.2 = /usr/lib/libxml2.so.2 (0xb746b000) libsqlite3.so.0 = /usr/lib/libsqlite3.so.0 (0xb73df000) libdl.so.2 = /lib/i686/cmov/libdl.so.2 (0xb73db000) libpthread.so.0 = /lib/i686/cmov/libpthread.so.0 (0xb73c2000) libncurses.so.5 = /lib/libncurses.so.5 (0xb7387000) libm.so.6 = /lib/i686/cmov/libm.so.6 (0xb7361000) libresolv.so.2 = /lib/i686/cmov/libresolv.so.2 (0xb734d000) libz.so.1 = /usr/lib/libz.so.1 (0xb7339000) /lib/ld-linux.so.2 (0xb7887000) # ldd /usr/lib/libspandsp.so linux-gate.so.1 = (0xb77a1000) libtiff.so.4 = /usr/lib/libtiff.so.4 (0xb769b000) libm.so.6 = /lib/i686/cmov/libm.so.6 (0xb7675000) libc.so.6 = /lib/i686/cmov/libc.so.6 (0xb752e000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0xb750e000) libz.so.1 = /usr/lib/libz.so.1 (0xb74fa000) /lib/ld-linux.so.2 (0xb77a2000) So, with those 2 commands, I couldn't directly check the link between asterisk and spandsp, and check am I'm really using spandsp0.0.6pre18. 2012/1/16 Olivier oza_4...@yahoo.fr Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 to check currently used libraries from command line ?
On Monday 16 January 2012, Olivier wrote: Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). To find out which libraries a particular binary executable program is linked against, you just need to do $ ldd /path/to/executable You can find the actual path to an executable by typing $ which foo Replace foo by the name of the executable about which you want information, obviously. Now, because we usually want the computer to do as much of the hard work for us as possible, we can use the $(command) operator -- which treats whatever is between the brackets as a command, runs it and substitutes its output into the command which it was part of -- to combine these two commands into one: $ ldd $(which foo) i.e. it will run which foo, and then do ldd on whatever output which foo returned. Trivial example below: $ ldd $(which ls) linux-vdso.so.1 = (0x7fffa7974000) libselinux.so.1 = /lib/libselinux.so.1 (0x7fdc03188000) librt.so.1 = /lib/librt.so.1 (0x7fdc02f8) libacl.so.1 = /lib/libacl.so.1 (0x7fdc02d78000) libc.so.6 = /lib/libc.so.6 (0x7fdc02a17000) libdl.so.2 = /lib/libdl.so.2 (0x7fdc02813000) /lib64/ld-linux-x86-64.so.2 (0x7fdc033ce000) libpthread.so.0 = /lib/libpthread.so.0 (0x7fdc025f6000) libattr.so.1 = /lib/libattr.so.1 (0x7fdc023f2000) Hmm, that's a whole lot of libraries just to get a directory listing! Not surprisingly, busybox gets away with rather less: $ ldd $(which busybox) linux-vdso.so.1 = (0x7fff2b7ff000) libm.so.6 = /lib/libm.so.6 (0x7f04a8217000) libc.so.6 = /lib/libc.so.6 (0x7f04a7eb6000) /lib64/ld-linux-x86-64.so.2 (0x7f04a84c1000) Note1: Here we see 16 hex digits after each library name, indicating a 64-bit system. On a 32-bit system, we would see only 8 hex digits after each library name. Note2: Programs that sometimes or always crash, may be missing a library. If so, this will be obvious when you run ldd. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real T1 trunk group...
Hey all! I'm not sure if this went out the first time I sent it so I apologize now if it's a duplicate. I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I migrated our incomming T1's from the Option 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance T1's. The local T1 went over with out a hitch. The problem is with my 2 long distance T1's. The switch on the other end is a DMS250 I'm told so I set Asterisk to DMS100 and got the timing, framing, etc all set. Well, the D channels came up so thats good. I started getting dropped calls every once in a while. I did a debug on the spans and saw the following: PRI Span: 3 PRI Span: 3 Protocol Discriminator: Q.931 (8) len=40 PRI Span: 3 TEI=0 Call Ref: len= 2 (reference 857/0x359) (Sent from originator) PRI Span: 3 Message Type: SETUP (5) PRI Span: 3 [04 03 80 90 a2] PRI Span: 3 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) PRI Span: 3 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 3 User information layer 1: u-Law (34) PRI Span: 3 [18 04 e9 80 83 08] PRI Span: 3 Channel ID (len= 6) [ Ext: 1 IntID: Explicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 3ChanSel: As indicated in following octets *PRI Span: 3Ext: 1 DS1 Identifier: 0 *PRI Span: 3Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 3Ext: 0 Channel: 8 Type: CPE] PRI Span: 3 [20 02 00 e2] PRI Span: 3 Network-Specific Facilities (len= 2) [ Toll Free MEGACOM ] PRI Span: 3 [6c 0c 21 83 37 32 37 34 3033 34 30 37 34] PRI Span: 3 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) The key part is the *Ext: 1 DS1 Identifier: 0* part. That's when calls fail. Right now, all calls are coming in on span 3 and want to talk to Identifier 0 (span 2). If a call comes in on span 2 and requests *Ext: 1 DS1 Identifier: 1*, it fails. I called Verizon and asked them what was going on. Turns out, its configured as a trunk group. The tech mentioned that I need to figure out how to set my identifiers on the group and I should be good to go. I've done a ton of research about chan_dahdi.conf and dahdi-channels.conf and I think the answer is trunk groups. I tried configuring a trunkgroup and set the primary dch to 24 and the bdch to 72 and then then spanmap'ed span 2 and 3 into group 1 (e.g. 2,1,0 and 3,1,1) but I don't see anything when I do a dahdi show channels or a pri show spans or a pri show channels, not even the channels not in the group. If I delete the trunkgroup, all three commands return all the channels. I'm just curious if I'm going down the right path with trunkgroups for this or if there is something else to take care of the DS1 Identifier issue. So another quick look... when a sucessful call comes in it goes to DS1 Identifier 0... the Asterisk CLI shows the following: -- Accepting call from '727403' to '890' on channel 0/11x-apple-data-detectors://0, span 2 Is there a way to get the other span (span 3) to become channel 1/xx? So when a call comes in asking for DS1 Identifier 1 I see the following: -- Accepting call from '727403' to '890' on channel 1/12, span 3 Thanks in advance everyone! Louis -- _ -- 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] Real T1 trunk group...
On 01/16/2012 04:48 AM, Louis Carreiro wrote: I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I migrated our incomming T1's from the Option 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance T1's. The local T1 went over with out a hitch. The problem is with my 2 long distance T1's. The switch on the other end is a DMS250 I'm told so I set Asterisk to DMS100 and got the timing, framing, etc all set. Well, the D channels came up so thats good. I started getting dropped calls every once in a while. I did a debug on the spans and saw the following: Sounds similar to what I am doing. Migrating from a Nortel Option 61 to Asterisk. I have everything set, and am just waiting for an appropriate window to move the 4 T-1s (2 trunk groups). All PRIs are national though, not DMS100. I tried configuring a trunkgroup and set the primary dch to 24 and the bdch to 72 and then then spanmap'ed span 2 and 3 into group 1 (e.g. 2,1,0 and 3,1,1) but I don't see anything when I do a dahdi show channels or a pri show spans or a pri show channels, not even the channels not in the group. If I delete the trunkgroup, all three commands return all the channels. I'm just curious if I'm going down the right path with trunkgroups for this or if there is something else to take care of the DS1 Identifier issue. Here are the relevant portions of my configs, I based them on a working model for PRIs connecting the Asterisk to the Option 61 as TIE trunks. This config has two dual port cards, with span 1 and 3 being a group and 2 and 4 being a different group. I hope this helps. (Or perhaps identifies something I have wrong that may not have been found yet ;-) Dale ### /etc/dahdi/system.conf ### # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF RED span=1,1,0,esf,b8zs # termtype: unknown bchan=1-23 dchan=24 echocanceller=hwec,1-23 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF RED span=2,2,0,esf,b8zs # termtype: unknown bchan=25-47 dchan=48 echocanceller=hwec,25-47 # Span 3: TE2/1/1 T2XXP (PCI) Card 1 Span 1 B8ZS/ESF RED span=3,3,0,esf,b8zs # termtype: unknown bchan=49-71 dchan=72 echocanceller=hwec,49-71 # Span 4: TE2/1/2 T2XXP (PCI) Card 1 Span 2 (MASTER) B8ZS/ESF RED span=4,4,0,esf,b8zs # termtype: unknown bchan=73-95 dchan=96 echocanceller=hwec,73-95 ## /etc/asterisk/chan_dahdi.conf ## [trunkgroups] trunkgroup = 1,24,72 trunkgroup = 2,48,96 spanmap = 1,1,0 spanmap = 3,1,1 spanmap = 2,2,0 spanmap = 4,2,1 [channels] ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF RED ; General Trunking group=1 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF ; IVR Trunking group=2 context=from-pstn switchtype = national signalling = pri_cpe channel = 25-47 ; Span 3: TE2/1/1 T2XXP (PCI) Card 1 Span 1 B8ZS/ESF ; General Trunking group=1 context=from-pstn switchtype = national signalling = pri_cpe channel = 49-71 ; Span 4: TE2/1/2 T2XXP (PCI) Card 1 Span 2 (MASTER) B8ZS/ESF RED ; IVR Trunking group=2 context=from-pstn switchtype = national signalling = pri_cpe channel = 73-95 -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 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] Peer doesn't answer
It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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] Peer doesn't answer
Paste some SIP traces of the call while Unmonitored. On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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] Real T1 trunk group...
Dale, That's funny! That is almost exactly what I'm trying to do. Thanks for the quick response! I'm on the way into the office now and I'll give the configuration a shot. I hope the config really helps. Maybe with our two migrations happening at the same time we maybe able to help each other out. I'll reply back within the hour! Louis Sent from my iPhone On Jan 16, 2012, at 6:59 AM, Dale Noll dn...@wi.rr.com wrote: On 01/16/2012 04:48 AM, Louis Carreiro wrote: I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I migrated our incomming T1's from the Option 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance T1's. The local T1 went over with out a hitch. The problem is with my 2 long distance T1's. The switch on the other end is a DMS250 I'm told so I set Asterisk to DMS100 and got the timing, framing, etc all set. Well, the D channels came up so thats good. I started getting dropped calls every once in a while. I did a debug on the spans and saw the following: Sounds similar to what I am doing. Migrating from a Nortel Option 61 to Asterisk. I have everything set, and am just waiting for an appropriate window to move the 4 T-1s (2 trunk groups). All PRIs are national though, not DMS100. I tried configuring a trunkgroup and set the primary dch to 24 and the bdch to 72 and then then spanmap'ed span 2 and 3 into group 1 (e.g. 2,1,0 and 3,1,1) but I don't see anything when I do a dahdi show channels or a pri show spans or a pri show channels, not even the channels not in the group. If I delete the trunkgroup, all three commands return all the channels. I'm just curious if I'm going down the right path with trunkgroups for this or if there is something else to take care of the DS1 Identifier issue. Here are the relevant portions of my configs, I based them on a working model for PRIs connecting the Asterisk to the Option 61 as TIE trunks. This config has two dual port cards, with span 1 and 3 being a group and 2 and 4 being a different group. I hope this helps. (Or perhaps identifies something I have wrong that may not have been found yet ;-) Dale ### /etc/dahdi/system.conf ### # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF RED span=1,1,0,esf,b8zs # termtype: unknown bchan=1-23 dchan=24 echocanceller=hwec,1-23 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF RED span=2,2,0,esf,b8zs # termtype: unknown bchan=25-47 dchan=48 echocanceller=hwec,25-47 # Span 3: TE2/1/1 T2XXP (PCI) Card 1 Span 1 B8ZS/ESF RED span=3,3,0,esf,b8zs # termtype: unknown bchan=49-71 dchan=72 echocanceller=hwec,49-71 # Span 4: TE2/1/2 T2XXP (PCI) Card 1 Span 2 (MASTER) B8ZS/ESF RED span=4,4,0,esf,b8zs # termtype: unknown bchan=73-95 dchan=96 echocanceller=hwec,73-95 ## /etc/asterisk/chan_dahdi.conf ## [trunkgroups] trunkgroup = 1,24,72 trunkgroup = 2,48,96 spanmap = 1,1,0 spanmap = 3,1,1 spanmap = 2,2,0 spanmap = 4,2,1 [channels] ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF RED ; General Trunking group=1 context=from-pstn switchtype = national signalling = pri_cpe channel = 1-23 ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF ; IVR Trunking group=2 context=from-pstn switchtype = national signalling = pri_cpe channel = 25-47 ; Span 3: TE2/1/1 T2XXP (PCI) Card 1 Span 1 B8ZS/ESF ; General Trunking group=1 context=from-pstn switchtype = national signalling = pri_cpe channel = 49-71 ; Span 4: TE2/1/2 T2XXP (PCI) Card 1 Span 2 (MASTER) B8ZS/ESF RED ; IVR Trunking group=2 context=from-pstn switchtype = national signalling = pri_cpe channel = 73-95 -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 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 -- _ -- 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] Peer doesn't answer
basically CLI shows SIP/X called SIP/Y I answer the call on Y but X keeps ringing and then both hangup. On Mon, Jan 16, 2012 at 8:01 AM, Sammy Govind govoi...@gmail.com wrote: Paste some SIP traces of the call while Unmonitored. On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 to check currently used libraries from command line ?
On Mon, Jan 16, 2012 at 11:14:48AM +0100, Olivier wrote: Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). To see the actual memory map of the process: pmap $PID_OF_ASTERISK Code is mapped from files, and thus you'll see the original files. You'll probably need to remove duplicates and such. Note that ldd of /usr/sbin/asterisk will not give you libraries of the various modules. For that you'll have to run ldd on the specific modules. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where to find meaning of /n in Local/6613@from-queue/n ?
Hi, Where to find meaning of /n in Local/6613@from-queue/n ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as UAC: How to put call OnHold
Ok, I will try this and let you know! Kind regards, John 2012/1/16 Sammy Govind govoi...@gmail.com: Hey, I have never worried about looking at the SIP re-invites or anything when we engage MoH() application in asterisk. You can do a quick test on your test machine for this. Regards, Sammy On Mon, Jan 16, 2012 at 2:57 PM, Johannes Zweng john999...@zweng.at wrote: Hi! Many thanks for this hint. I will try this! :-) A quick question: when doing this with MusicOnHold(): will the SIP server be aware that the call is placed onHold (i.e. will Asterisk send the mentioned re-INVITE)? The point is - if possible - we want the caller to hear the OnHold Music from the SIP server. If not we would have to copy the MoH to our Asterisk (and change it on our side too, when it changes at the SIP-server). Kind regards, John 2012/1/16 Sammy Govind govoi...@gmail.com Hi, yes, please see MusicOnHold() Application. You can call this app in your dialplan. This however will use the default music class and the corresponding music files placed in the asterisk server. If you don't want to stream music from Asterisk server side, try creating a new MusiconHold Class without any proper directory. That way Asterisk would only complain that there is no file to be streamed. Regards, Sammy On Sat, Jan 14, 2012 at 6:25 AM, Johannes Zweng john999...@zweng.at wrote: Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: -- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the dialplan and should place the call on hold after some time, so that the caller only hears the on hold music from my provider (not streamed by my Asterisk). Technically speaking I want asterisk to send a re-INVITE message containing an updated SDP body with the attribute a=sendonly or a=inactive added so that the SIP server of my provider (where Asterisk is registered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) (http://tools.ietf.org/html/rfc5359#section-2.1) where Bob would be my Asterisk (as UAC), Alice is the external caller and Proxy is the provider's SIP server. Question: -- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like Hold? Kind regards and greetings from Austria, John :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to find meaning of /n inLocal/6613@from-queue/n ?
http://www.voip-info.org/wiki/view/Asterisk+local+channels Regards - Original Message - From: Olivier oza_4...@yahoo.fr To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 16, 2012 7:41 AM Subject: [asterisk-users] Where to find meaning of /n inLocal/6613@from-queue/n ? Hi, Where to find meaning of /n in Local/6613@from-queue/n ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Where to find meaning of /n inLocal/6613@from-queue/n ? [SOLVED]
2012/1/16, bakko asannu...@gmail.com: http://www.voip-info.org/wiki/view/Asterisk+local+channels I don't know why but I was thinking of some sort Dial app magic and didn't look after Local channels options. Thanks for correcting me. Regards - Original Message - From: Olivier oza_4...@yahoo.fr To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 16, 2012 7:41 AM Subject: [asterisk-users] Where to find meaning of /n inLocal/6613@from-queue/n ? Hi, Where to find meaning of /n in Local/6613@from-queue/n ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Peer doesn't answer
Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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] Peer doesn't answer
the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda flaviormira...@hotmail.comwrote: Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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] Peer doesn't answer
I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense. On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda flaviormira...@hotmail.com wrote: Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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] Exceptionally long voice queue length
Anybody? I've read this might be a deadlock On Thu, Jan 12, 2012 at 8:09 AM, Vik Killa vipki...@gmail.com wrote: Asterisk 1.6.1.22 On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind govoi...@gmail.com wrote: which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! -- _ -- 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] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via
a menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are
being played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing
'main-menu.gsm' (language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
with asterisk
;nat=yes; assume the
device is behind nat
secret=xxx ; a secure password for this device
dtmfmode=auto ; accept touch-tones from
devices, negotiated automatically
disallow=all; reset with voice codecs
to accept from, and request to, the device
allow=alaw ; which audio codecs we
accept from
canreinvite=nonat
--
_
-- 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] SayDigits playback doesn't always work
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before the
SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but it
should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via a
menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are being
played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm'
(language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
with asterisk
;nat=yes; assume the device
is behind nat
secret=xxx ; a secure password for this device
dtmfmode=auto ; accept touch-tones from
devices, negotiated automatically
disallow=all; reset with voice codecs to
accept from, and request to, the device
allow=alaw ; which audio codecs we
accept from
canreinvite=nonat
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Re: [asterisk-users] SayDigits playback doesn't always work
Ok, got it. Indeed, starting with Answer() helped.
But I still don't understand why the parking feature isn't working then. I
used the sample config. Transfer the call to 700, playback of the lot is
being executed, but I hear nothing. Probably the same problem, but how do I
change this?
On Mon, Jan 16, 2012 at 4:26 PM, Danny Nicholas da...@debsinc.com wrote:
You aren’t “opening the line” in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are
“plunging” into the SayDigits() function without opening the channel. Some
functions will generate their own Answer() if not present, others will not.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Roland
*Sent:* Monday, January 16, 2012 9:22 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] SayDigits playback doesn't always work
** **
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
** **
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be played back to the one who is parking of course.
** **
So something seems to be wrong with SayDigits?
** **
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:***
*
Hi,
** **
I have this wierd problem where SayDigits does work when I execute it via
a menu, but not when calling directly. In my extensions, I have this setup:
** **
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
** **
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
** **
** **
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
** **
When I call 123 from the same phone, I do see that the sound files are
being played to me, but I don't hear any sound.
** **
In Asterisk CLI I see this:
** **
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
** **
This is the call that doesn't work. Then when I call 200, I see this:
** **
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing
'main-menu.gsm' (language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
** **
This call works perfectly. What am I missing?
** **
In my sip.conf I have:
** **
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the
[asterisk-users] How Can I configure the between call oneside IVR
Hi list, how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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] SayDigits playback doesn't always work
Post your dialplan snippet you use to park the call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
Ok, got it. Indeed, starting with Answer() helped.
But I still don't understand why the parking feature isn't working then. I
used the sample config. Transfer the call to 700, playback of the lot is
being executed, but I hear nothing. Probably the same problem, but how do I
change this?
On Mon, Jan 16, 2012 at 4:26 PM, Danny Nicholas da...@debsinc.com wrote:
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before the
SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but it
should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via a
menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are being
played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm'
(language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
Re: [asterisk-users] SayDigits playback doesn't always work
This symptom usually means you are doing an attended transfer instead of a blind transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland Sent: Monday, January 16, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SayDigits playback doesn't always work Ok, got it. Indeed, starting with Answer() helped. But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this? This is the call that doesn't work. Then when I call 200, I see this: [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1 [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer(SIP/000B822FD265-003e, ) in new stack [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround(SIP/000B822FD265-003e, main-menu) in new stack [Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm' (language 'nl') [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten(SIP/000B822FD265-003e, 5) in new stack [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait(SIP/000B822FD265-003e, 2) in new stack [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits(SIP/000B822FD265-003e, 123) in new stack [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm' (language 'nl') [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm' (language 'nl') [Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm' (language 'nl') [Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-003e' status is 'UNKNOWN' [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1 This call works perfectly. What am I missing? In my sip.conf I have: [stumpel-zwaag](!) ; create template for our devices type=friend ; the channel driver will mathc on username first, IP second context=StumpelZwaag; this is where calls from the device will enter the dialplan host=dynamic; the device will register with asterisk ;nat=yes; assume the device is behind nat secret=xxx ; a secure password for this device dtmfmode=auto ; accept touch-tones from devices, negotiated automatically disallow=all; reset with voice codecs to accept from, and request to, the device allow=alaw ; which audio codecs we accept from canreinvite=nonat -- _ -- 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] How Can I configure the between call oneside IVR
A should transfer C to a local channel that plays the IVR then returns the call to A. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta Sent: Monday, January 16, 2012 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How Can I configure the between call oneside IVR Hi list, how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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] Update callee num or name at caller display
Hi, A calls B and B has it's phone forwarded to C. So the call rings at C. Is there any way to inform A about that forwarding? Best way would be to update the called name so A has B forwarded to C in his display. Any chance to get this? I tried Set(REDIRECTING(to-name)=...). This sends a SIP/2.0 181 Call is being forwarded to the calling phone, but with no information about the new callee name. Regards, Gunnar -- _ -- 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] SayDigits playback doesn't always work
I am just starting with Asterisk .. I think you are right, I am doing an attended transfer, although I don't exactly understand what that means. I still need to know in what lot I can pickup my call again right? Ok, my config .. (i will leave out the commented stuff, because there's lot of comments in the sample config) [general] parkext = 700 ; What extension to dial to park. Set per parking lot. parkpos = 701-720 ; What extensions to park calls on. (defafult parking lot) context = parkedcalls ; Which context parked calls are in (default parking lot) parkingtime = 300 ; Number of seconds a call can be parked before returning. comebacktoorigin = yes ; Setting this option configures the behavior of call parking when the courtesytone = beep; Sound file to play to when someone picks up a parked call parkedplay = both; Who to play courtesytone to when picking up a parked call. Thanks! On Mon, Jan 16, 2012 at 4:59 PM, Eric Wieling ewiel...@nyigc.com wrote: This symptom usually means you are doing an attended transfer instead of a blind transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Roland Sent: Monday, January 16, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SayDigits playback doesn't always work Ok, got it. Indeed, starting with Answer() helped. But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this? This is the call that doesn't work. Then when I call 200, I see this: [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1 [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer(SIP/000B822FD265-003e, ) in new stack [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround(SIP/000B822FD265-003e, main-menu) in new stack [Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm' (language 'nl') [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten(SIP/000B822FD265-003e, 5) in new stack [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait(SIP/000B822FD265-003e, 2) in new stack [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits(SIP/000B822FD265-003e, 123) in new stack [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm' (language 'nl') [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm' (language 'nl') [Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm' (language 'nl') [Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-003e' status is 'UNKNOWN' [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1 This call works perfectly. What am I missing? In my sip.conf I have: [stumpel-zwaag](!) ; create template for our devices type=friend ; the channel driver will mathc on username first, IP second context=StumpelZwaag; this is where calls from the device will enter the dialplan host=dynamic; the device will register with asterisk ;nat=yes; assume the device is behind nat secret=xxx ; a secure password for this device dtmfmode=auto ; accept touch-tones from devices, negotiated automatically disallow=all; reset with voice codecs to accept from, and request to, the device allow=alaw ; which audio codecs we accept from canreinvite=nonat -- _ -- 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
Re: [asterisk-users] How Can I configure the between call oneside IVR
I was tried it but its not going.. with same Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas da...@debsinc.com wrote: A should transfer C to a local channel that plays the IVR then returns the call to A. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mahesh katta *Sent:* Monday, January 16, 2012 9:56 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] How Can I configure the between call oneside IVR ** ** Hi list, how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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] How Can I configure the between call oneside IVR
I would do it something like this
[ivrandreturn]
Exten = s,1,playback(message)
Exten = s,n,waitexten(5)
Exten = 1,1,noop(stuff for press 1)
Exten = 1,n,dial(SIP/A)
Exten = 2,1,noop(stuff for press 2)
Exten = 2,n,dial(SIP/A)
In real life SIP/A would be something like SIP/${ARG1} where ARG1 is the
extension for A.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Monday, January 16, 2012 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How Can I configure the between call oneside
IVR
I was tried it but its not going.. with same
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas da...@debsinc.com wrote:
A should transfer C to a local channel that plays the IVR then returns the
call to A.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Monday, January 16, 2012 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How Can I configure the between call oneside IVR
Hi list,
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play for C, C enter the some DTMF inputs and A should be on hold.
after finish C input will complete again they want talk each other .This
is the scenario.
Can anybody help to me how can I add this IVR in between those call, and
how my asterisk will detect the DTMF input
Best Regards,
Mahesh Katta
--
_
-- 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] Problems with codec translation when using Monitor and MixMonitor
Yes, a 'call' refers to two channels bridged. Jim, please help me to undertand the numbers. I have two g729 licenses, my SIP provider uses only g729 and my softphones support g729 too, asterisk.conf is set in its default value (sln). When a call (2 channels) is being made and succesfully recorded with MixMonitor (wav49 format), I see at CLI: testpbx*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer A.B.C.D 987654321639237c5976 0x100 (g729) No Tx: ACKsip-provider1 W.X.Y.Z elder 4e4adc85-b2e21c0x100 (g729) No Rx: ACKelder testpbx*CLI g729 show licenses 0/2 encoders/decoders of 2 licensed channels are currently in use Licenses Found: File: G729-... -- Key: G729-...-- Host-ID: ... -- Channels: 1 (Expires: 20...) (OK) File: G729-... -- Key: G729-...-- Host-ID: ... -- Channels: 1 (Expires: 20...) (OK) Thanks for your answers, Elder On Thu, Jan 12, 2012 at 6:05 PM, Jim Dickenson dicken...@cfmc.com wrote: Here is a matrix we put together about g729 license needs: == = == === Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders == = == === ulaw ulaw yes yesyes00 ulaw ulaw yes yesno 00 ulaw ulaw yes no no 00 ulaw ulaw yes no yes00 ulaw ulaw no yesyes00 ulaw ulaw no yesno 00 ulaw ulaw no no no 00 ulaw ulaw no no yes00 ulaw g729 yes yesyes33 ulaw g729 yes yesno 23 ulaw g729 yes no no 11 ulaw g729 yes no yes33 ulaw g729 no yesyes33 ulaw g729 no yesno 23 ulaw g729 no no no 11 ulaw g729 no no yes33 g729 ulaw yes yesyes25 g729 ulaw yes yesno 25 g729 ulaw yes no no 11 g729 ulaw yes no yes23 g729 ulaw no yesyes25 g729 ulaw no yesno 25 g729 ulaw no no no 11 g729 ulaw no no yes23 g729 g729 yes yesyes47 g729 g729 yes yesno 37 g729 g729 yes no no 11 g729 g729 yes no yes45 g729 g729 no yesyes47 g729 g729 no yesno 37 g729
Re: [asterisk-users] How Can I configure the between call oneside IVR
Best Regards,
ahesh Katta
On Mon, Jan 16, 2012 at 9:57 PM, Danny Nicholas da...@debsinc.com wrote:
I would do it something like this
[ivrandreturn]
Exten = s,1,playback(message)
Exten = s,n,waitexten(5)
Exten = 1,1,noop(stuff for press 1)
Exten = 1,n,dial(SIP/A)
Exten = 2,1,noop(stuff for press 2)
Exten = 2,n,dial(SIP/A)
** **
In real life SIP/A would be something like SIP/${ARG1} where ARG1 is the
extension for A.
**
In this scenario A does not HOLD, its Disconnect, I need it should be
hold. it should be in conference.
**
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *mahesh katta
*Sent:* Monday, January 16, 2012 10:21 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How Can I configure the between call
oneside IVR
** **
I was tried it but its not going.. with same
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas da...@debsinc.com wrote:
A should transfer C to a local channel that plays the IVR then returns the
call to A.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *mahesh katta
*Sent:* Monday, January 16, 2012 9:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] How Can I configure the between call oneside
IVR
Hi list,
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play for C, C enter the some DTMF inputs and A should be on hold.
after finish C input will complete again they want talk each other .This
is the scenario.
Can anybody help to me how can I add this IVR in between those call,
and how my asterisk will detect the DTMF input
Best Regards,
Mahesh Katta
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Re: [asterisk-users] Where to find meaning of /n in Local/6613@from-queue/n ?
Where to find meaning of /n in Local/6613@from-queue/n ? See https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers Richard -- _ -- 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] meetme with IVR
Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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] Starting things off without a dial tone
Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a dialling tone and waiting for the user to dial an extension? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting things off without a dial tone
On Mon, Jan 16, 2012 at 05:52:10PM +, A J Stiles wrote: Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a dialling tone and waiting for the user to dial an extension? With analog phones (chan_dahdi) - 'immediate = yes' in chan_dahdi.conf . With a SIP phone: that's something to configure the handset for, as it only sends out a call once you dialed. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Hello, I can do simple, yum install asterisk18-* and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *You do not appear to have the source for the 2.6.32-4-pve kernel installed.* * * 1- Based on above error and Google search I have concluded that dahdi-linux module should be installed on mother node. So, I am puzzled. How does Digium yum repository achive this without acessing the mother node? 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Any feedback is much appreciated. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
On Mon, Jan 16, 2012 at 01:41:30PM -0500, asterisk jobs wrote: 1- Based on above error and Google search I have concluded that dahdi-linux module should be installed on mother node. So, I am puzzled. How does Digium yum repository achive this without acessing the mother node? The repo files are pre-compiled and do not require the kernel headers. If you wish to compile dahdi from source, you'll need access to the same headers your VM is running. 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). Regards, Patrick -- _ -- 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] Update callee num or name at caller display
I've never done it myself yet but I think I would look after COLP function (1.8 and above). -- _ -- 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] Update callee num or name at caller display
On 01/16/2012 12:55 PM, Olivier wrote: I've never done it myself yet but I think I would look after COLP function (1.8 and above). Asterisk 1.8 and later will do this automatically; if the phone can display the redirection information, it will get displayed (not all phones can do so). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
On 01/16/2012 12:52 PM, Patrick Lists wrote: On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). They have changed; DAHDI is required for MeetMe/SLA/Page, but is not required for timing. In Asterisk 10, ConfBridge can be a suitable replacement for MeetMe for many users as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
2012-01-16 19:41, asterisk jobs skrev: Hello, I can do simple, yum install asterisk18-* and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, /*You do not appear to have the source for the 2.6.32-4-pve kernel installed.*/ /* */ 1- Based on above error and Google search I have concluded that dahdi-linux module should be installed on mother node. So, I am puzzled. How does Digium yum repository achive this without acessing the mother node? 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? I've just installed a new server with OpenVZ. And as others has explained you will need Dahdi for Meetme among other things. You will need to install dahdi-complete on the Hardware node, and the kernel sources. ( Debian: apt-get install linux-headers-`uname -r` ) In the VE conf-file you will need the following line for the VE to access Dahdi: DEVNODES=dahdi/channel:rw dahdi/ctl:rw dahdi/pseudo:rw dahdi/timer:rw In the VE, compile and install dahdi-complete, then build and install asterisk. -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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 does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Thanks for all the input guys. I am using Asterisk 1.8 for this purpose. 1- So, I do I still need Dahdi? And yes conference will be used. 2- Can you please detail on compiled already code? My mother node for OpenVz is probably different from what Digium uses to compile the source. How does this work? 3- How can I compile my own source code and then move it to my OpenVZ to work just the same? Thanks again On Mon, Jan 16, 2012 at 1:57 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/16/2012 12:52 PM, Patrick Lists wrote: On 16-01-12 19:47, Russ Meyerriecks wrote: [snip] 2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and it's all SIP? If yes, what do I need it for? Dahdi is a set of drivers for telephony hardware. You won't need it for pure sip Asterisk implementations. Unless things have changed with recent versions I think you still need DAHDI if you want to use MeetMe and maybe other modules that require proper timing (which DAHDI provides). They have changed; DAHDI is required for MeetMe/SLA/Page, but is not required for timing. In Asterisk 10, ConfBridge can be a suitable replacement for MeetMe for many users as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update callee num or name at caller display
On 01/16/2012 12:55 PM, Olivier wrote: I've never done it myself yet but I think I would look after COLP function (1.8 and above). Asterisk 1.8 and later will do this automatically; if the phone can display the redirection information, it will get displayed (not all phones can do so). Thanks for the feedback. Any documentation abount COLP? On voip-info.org there is noting. The redirection is done in Asterisk dialplan, so I have to tell phone A about the forwarding. exten = B,1,Dial(SIP/C) So I need a dialplan function or something else to send an update to phone A. Regards, Gunnar -- _ -- 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] Update callee num or name at caller display
See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special attention to the sendrpid note. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gunnar Schaller Sent: Monday, January 16, 2012 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Update callee num or name at caller display On 01/16/2012 12:55 PM, Olivier wrote: I've never done it myself yet but I think I would look after COLP function (1.8 and above). Asterisk 1.8 and later will do this automatically; if the phone can display the redirection information, it will get displayed (not all phones can do so). Thanks for the feedback. Any documentation abount COLP? On voip-info.org there is noting. The redirection is done in Asterisk dialplan, so I have to tell phone A about the forwarding. exten = B,1,Dial(SIP/C) So I need a dialplan function or something else to send an update to phone A. Regards, Gunnar -- _ -- 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] Update callee num or name at caller display
Hello Eric, See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special attention to the sendrpid note. That does not work. CONNECTEDLINE is for answered calls. A calls B. B has a forward to C in Asterisk dialplan. A want's to notice the forwarding _before_ C answers. Cause A only want to speak to B. Sorry if that was not clear before. Regards, Gunnar -- _ -- 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] OT - Configuring Freepbx's fax_process.pl to work with ssmtp
Hi, Freepbx includes a fax_process.pl which convert TIF files into PDF files before sending by email. I'm used to use sSMTP with Asterisk. I'm certain ssmtp is correctly configured in my (Debian Squeeze) setup as I'm correctly receiving voicemails in email box. Is it possible to tell fax_process.pl to use ssmtp when sending emails out ? If positive, any hint on how to configure this ? If negative, which smtp software shall I replace ssmtp with ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update callee num or name at caller display
Are both A and B extensions of the same Asterisk system or is A an inbound caller ? 2012/1/16, Gunnar Schaller li...@nowin.de: Hello Eric, See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special attention to the sendrpid note. That does not work. CONNECTEDLINE is for answered calls. A calls B. B has a forward to C in Asterisk dialplan. A want's to notice the forwarding _before_ C answers. Cause A only want to speak to B. Sorry if that was not clear before. Regards, Gunnar -- _ -- 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] Real T1 trunk group...
On Mon, Jan 16, 2012 at 5:48 AM, Louis Carreiro carreir...@gmail.com wrote: Hey all! I'm not sure if this went out the first time I sent it so I apologize now if it's a duplicate. I've been banging my head against the wall for a while (almost 18 hours today alone) with this one... I migrated our incomming T1's from the Option 11 to our Asterisk box this morning. We have 1 local T1 and 2 long distance T1's. The local T1 went over with out a hitch. The problem is with my 2 long distance T1's. The switch on the other end is a DMS250 I'm told so I set Asterisk to DMS100 and got the timing, framing, etc all set. Well, the D channels came up so thats good. I started getting dropped calls every once in a while. I did a debug on the spans and saw the following: I have found that in most cases the easiest way to fix these issues is to simply call the provider and ask them to switch it to NI2. Most of them can do that while on the phone. PRI Span: 3 PRI Span: 3 Protocol Discriminator: Q.931 (8) len=40 PRI Span: 3 TEI=0 Call Ref: len= 2 (reference 857/0x359) (Sent from originator) PRI Span: 3 Message Type: SETUP (5) PRI Span: 3 [04 03 80 90 a2] PRI Span: 3 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) PRI Span: 3 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 3 User information layer 1: u-Law (34) PRI Span: 3 [18 04 e9 80 83 08] PRI Span: 3 Channel ID (len= 6) [ Ext: 1 IntID: Explicit Other(PRI) Spare: 0 Exclusive Dchan: 0 PRI Span: 3 ChanSel: As indicated in following octets PRI Span: 3 Ext: 1 DS1 Identifier: 0 PRI Span: 3 Ext: 1 Coding: 0 Number Specified Channel Type: 3 PRI Span: 3 Ext: 0 Channel: 8 Type: CPE] PRI Span: 3 [20 02 00 e2] PRI Span: 3 Network-Specific Facilities (len= 2) [ Toll Free MEGACOM ] PRI Span: 3 [6c 0c 21 83 37 32 37 34 3033 34 30 37 34] PRI Span: 3 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) The key part is the Ext: 1 DS1 Identifier: 0 part. That's when calls fail. Right now, all calls are coming in on span 3 and want to talk to Identifier 0 (span 2). If a call comes in on span 2 and requests Ext: 1 DS1 Identifier: 1, it fails. I called Verizon and asked them what was going on. Turns out, its configured as a trunk group. The tech mentioned that I need to figure out how to set my identifiers on the group and I should be good to go. I've done a ton of research about chan_dahdi.conf and dahdi-channels.conf and I think the answer is trunk groups. I tried configuring a trunkgroup and set the primary dch to 24 and the bdch to 72 and then then spanmap'ed span 2 and 3 into group 1 (e.g. 2,1,0 and 3,1,1) but I don't see anything when I do a dahdi show channels or a pri show spans or a pri show channels, not even the channels not in the group. If I delete the trunkgroup, all three commands return all the channels. I'm just curious if I'm going down the right path with trunkgroups for this or if there is something else to take care of the DS1 Identifier issue. So another quick look... when a sucessful call comes in it goes to DS1 Identifier 0... the Asterisk CLI shows the following: -- Accepting call from '727403' to '890' on channel 0/11, span 2 Is there a way to get the other span (span 3) to become channel 1/xx? So when a call comes in asking for DS1 Identifier 1 I see the following: -- Accepting call from '727403' to '890' on channel 1/12, span 3 Thanks in advance everyone! Louis -- _ -- 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] meetme with IVR
Any one is help ? Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.comwrote: Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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] Update callee num or name at caller display
Are both A and B extensions of the same Asterisk system or is A an inbound caller ? Both are snom phones at the same Asterisk (1.8.8). Regards, Gunnar -- _ -- 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
