[asterisk-users] asterisk speech to text and text to speech?
Dear All Can you please let me know if the asterisk has speech to text and text to speech facilities? Thank you -- _ -- 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 stop Dial from waiting for extra digits if number is incomplete.
We have Asterisk 1.6.2.17.2 connected to a sangoma E1 card. The problem
we are having is that we have a calling card type application and when
people enter the number to be dialled we call the Dial application. It
gets back an indication that the number is incomplete (via PRI cause
code 28 I assume) and waits for extra digits. If it gets extra digits it
appends them to the current extension and jumps to priority 1.
Is there any way this behaviour can be changed so if Dial tries dialling
an incomplete number it just fails and we can query ${HANGUPCAUSE} to
decide what to do.
Thanks
Gareth
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[asterisk-users] MixMonitor and attended transfers
Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording only lasts up to the point that the transfer is made. Is this the correct behaviour? Is there any way I could make this inbound call into a single continuous recording? Thanks in advance Ish -- 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
Re: [asterisk-users] MixMonitor and attended transfers
Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording only lasts up to the point that the transfer is made. Is this the correct behaviour? Is there any way I could make this inbound call into a single continuous recording? I've not used 1.8 yet, but in 1.4, you could send the incoming call through a LOCAL channel when the call comes in, and start the recording on the Local channel. That way, the LOCAL channel should keep recording, even when you transfer the call. You may need to add /n http://www.voip-info.org/wiki/view/Asterisk+local+channels Hope that helps. It's a little hard to explain, but try it out. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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] MixMonitor and attended transfers
On Tue, 2011-08-02 at 07:51 -0400, Dan Journo wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording only lasts up to the point that the transfer is made. Is this the correct behaviour? Is there any way I could make this inbound call into a single continuous recording? I've not used 1.8 yet, but in 1.4, you could send the incoming call through a LOCAL channel when the call comes in, and start the recording on the Local channel. That way, the LOCAL channel should keep recording, even when you transfer the call. You may need to add /n http://www.voip-info.org/wiki/view/Asterisk+local+channels Hope that helps. It's a little hard to explain, but try it out. Dan Journo Kesher Communications (UK) Business Phone Systems | Hosted PBX I didn't have this problem with 1.4, it just recorded the whole message as a matter of course. I'll have a look into using local channels for this but I think it has more to do with the way that 1.8 as treating attended transfers and how it joins the 2 channels involved. -- 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
Re: [asterisk-users] MixMonitor and attended transfers
On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording only lasts up to the point that the transfer is made. Is this the correct behaviour? Is there any way I could make this inbound call into a single continuous recording? Thanks in advance Ish Here's part of the log for this procedure [2011-08-02 13:47:13] VERBOSE[6475] rtp_engine.c: -- Locally bridging SIP/A-0049 and SIP/B-004a [2011-08-02 13:47:20] VERBOSE[6475] rtp_engine.c: -- Locally bridging SIP/inbound-0047 and SIP/B-004a [2011-08-02 13:47:20] VERBOSE[6463] pbx.c: == Spawn extension (inbound, s, 4) exited non-zero on 'SIP/A-0049ZOMBIE' [2011-08-02 13:47:20] VERBOSE[6464] app_mixmonitor.c: == MixMonitor close filestream [2011-08-02 13:47:26] VERBOSE[6475] app_macro.c: == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/inbound-0047' in macro 'stdexten' [2011-08-02 13:47:26] VERBOSE[6475] pbx.c: == Spawn extension (local, B, 1) exited non-zero on 'SIP/inbound-0047' [2011-08-02 13:47:26] VERBOSE[6464] app_mixmonitor.c: == End MixMonitor Recording SIP/inbound-0047 Obviously, I've obscured some of the more sensitive details in there The thing to notice here though is that MixMonitor closes the filestream when I hit the transfer button but actually Ends the recording 6 seconds later when the whole call was ended. This seems like inconsistent behaviour and more like an unintentional consequence of changes rather than intended behaviour, i.e. why would you close the filestream yet not end the recording? -- 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
[asterisk-users] SMS within asterisk users
I'm trying to setup SMS among users of a single asterisk box.
I've set up asterisk10-beta to send SMS messages using MessageSend(). If I
manually set the 'from' variable. I can two way messages only between those
two extensions.
i.e.
[sms]
exten = 12000,1,MessageSend(sip:12000,12001)
exten = 12001,1,MessageSend(sip:12001,12000)
This works fine, But limits the users to only be able to text each other
back and forth. When I add a third extension matters are complicated.
So I tried to set up something that was a little more flexible. I thought
that I would be able to use ${CALLERID(num)} for the 'from' variable.
[sms]
exten = _X.,1,MessageSend(${EXTEN},${CALLERID(num)})
However, the CALLERID(num) variable is an empty string.
Is there another way to identify the extension that originated the message?
--cobra2
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Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
I am happy it's being taken care of. Would the patch fix systems that used the Repo to install Asterisk 1.6.2.19? That is where we all have problems. Or maybe a new version of Asterisk which yum update would do the job? On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] use ILBC installed from asterisk yum repositories
I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 1.8.5.0 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? Thanks, Bob -- _ -- 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] use ILBC installed from asterisk yum repositories
On 08/02/2011 11:42 AM, Bob Pierce wrote: I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 1.8.5.0 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? There is no codec_ilbc RPM available from the Digium repositories at this point; there could be one in the future, but given that this is the first time I've seen a request for it, it seems unlikely to be worth the effort. You can use the SRPM for Asterisk to rebuild the RPM after importing the iLBC source into the build tree; at least I think that would work. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
Hi, I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy) I also compiled iksemel (v1.4) with the option 2./configure --with-libgnutls-prefix=/usr As explained in this link (to avoid compilation error ) http://code.google.com/p/iksemel/issues/detail?id=29#c3 I configured jabber.conf and gtalk.conf as explained in wiki.asterisk.org, but I have this error when starting : asterisk -c ... ... JABBER: asterisk INCOMING: stream:stream from=gmail.com id=57A17D99ADBC6814 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttlsmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features [Aug 2 12:57:40] ERROR[31479]: res_jabber.c:1610 aji_act_hook: OpenSSL not installed. You need to install OpenSSL on this system, or disable the TLS option in your configuration file [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:1391 aji_recv: Parsing failure: Hook returned an error. [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:2742 aji_recv_loop: JABBER: Got hook event. [Aug 2 12:57:56] WARNING[31479]: res_jabber.c:2753 aji_recv_loop: JABBER: socket read error But I have already openssl : root@mohamed-desktop:/usr/src# dpkg -l |grep -i openssl ii libcurl3 7.21.0-1ubuntu1.1 Multi-protocol file transfer library (OpenSSL) ii libxmlsec1-openss 1.2.14-1+squeeze1build0.10.10.1 Openssl engine for the XML security library ii openssl 0.9.8o-1ubuntu4.4 Secure Socket Layer (SSL) binary and related cryptographic tools ii python-op 0.10-1 Python wrapper around the OpenSSL library ii ssl-cert 1.0.26 simple debconf wrapper for OpenSSL Have you any idea where is the problem ? NB: I didn´t have that problem with asterisk 1.6 Thx -- _ -- 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] use ILBC installed from asterisk yum repositories
I would be very interested in iLBC. I even posted regarding this to this mailing list and the thread died after no one was able to confirm it works. I think there are others who would really like to see H.323 working from the repo as well (I think that is not working as well). Regards, On Tue, Aug 2, 2011 at 12:46 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 08/02/2011 11:42 AM, Bob Pierce wrote: I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 1.8.5.0 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? There is no codec_ilbc RPM available from the Digium repositories at this point; there could be one in the future, but given that this is the first time I've seen a request for it, it seems unlikely to be worth the effort. You can use the SRPM for Asterisk to rebuild the RPM after importing the iLBC source into the build tree; at least I think that would work. -- 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] use ILBC installed from asterisk yumrepositories
If we are talking about adding stuff to the repo I would vote for jabber and gtalk also fax (spandsp) -Original Message- From: Bruce B bruceb...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 2 Aug 2011 13:36:31 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] use ILBC installed from asterisk yum repositories -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec negotiation issue (no audio format found to offer)
Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound to my SIP trunk. Here are the basics of my config, showing the codec list from sip show peer peer: Polycom SP501 (desk phone): -- disallow=all allow=ulawg729 Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Grandstream HT503 (fxo gateway): -- disallow=all allow=ulawg729 Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) CallWithUs (SIP trunk): -- disallow=all allow=g729 Codecs : 0x100 (g729) Codec Order : (g729:20) When I place an outbound call from the Polycom to callwithus, the invite from the pcom shows both ulaw and g729 in the SDP: INVITE sip:@192.168.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKc8aa981a8B8FF58D From: Office sip:2001@192.168.0.1;tag=4CD2B2A0-B94A2531 To: sip:919785013620@192.168.0.1;user=phone [...] m=audio 2258 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Asterisk sees this: [Aug 2 15:00:31] VERBOSE[1918] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729) The call goes out the callwithus trunk: [Aug 2 15:00:31] VERBOSE[1315] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/2001-0047, SIP/CallWithUs/**,300,tTwW) in new stack And then this, no INVITE goes out to callwithus at all: [Aug 2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer. Cancelling call to ** [Aug 2 15:00:31] VERBOSE[1315] app_dial.c: -- Couldn't call SIP/CallWithUs/** Similarly, if I set the Grandstream FXO trunk to ulaw only, the call fails as well. It seems as if allowing only a single codec is the issue, if I change the priorities of all codecs to g729 first and ulaw second, the call goes through as g729 successfully. Smells like a bug to me, but I may be overlooking something in my config. Thanks, -Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
Install OpenSSL-devel (or whatever the equivalent ubuntu package is called) and then recompile / reinstall and test it again. Thanks, --Warren Selby, dCAP On Aug 2, 2011, at 12:06 PM, neo haux neo.h...@gmx.com wrote: Hi, I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy) I also compiled iksemel (v1.4) with the option 2./configure --with-libgnutls-prefix=/usr As explained in this link (to avoid compilation error ) http://code.google.com/p/iksemel/issues/detail?id=29#c3 I configured jabber.conf and gtalk.conf as explained in wiki.asterisk.org, but I have this error when starting : asterisk -c ... ... JABBER: asterisk INCOMING: stream:stream from=gmail.com id=57A17D99ADBC6814 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tlsrequired//starttlsmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanismmechanismX-OAUTH2/mechanism/mechanisms/stream:features [Aug 2 12:57:40] ERROR[31479]: res_jabber.c:1610 aji_act_hook: OpenSSL not installed. You need to install OpenSSL on this system, or disable the TLS option in your configuration file [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:1391 aji_recv: Parsing failure: Hook returned an error. [Aug 2 12:57:40] WARNING[31479]: res_jabber.c:2742 aji_recv_loop: JABBER: Got hook event. [Aug 2 12:57:56] WARNING[31479]: res_jabber.c:2753 aji_recv_loop: JABBER: socket read error But I have already openssl : root@mohamed-desktop:/usr/src# dpkg -l |grep -i openssl ii libcurl3 7.21.0-1ubuntu1.1 Multi-protocol file transfer library (OpenSSL) ii libxmlsec1-openss 1.2.14-1+squeeze1build0.10.10.1 Openssl engine for the XML security library ii openssl0.9.8o-1ubuntu4.4 Secure Socket Layer (SSL) binary and related cryptographic tools ii python-op 0.10-1Python wrapper around the OpenSSL library ii ssl-cert 1.0.26simple debconf wrapper for OpenSSL Have you any idea where is the problem ? NB: I didn´t have that problem with asterisk 1.6 Thx -- _ -- 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] use ILBC installed from asterisk yum repositories
On Tue, 2 Aug 2011 11:42:19 -0500 Bob Pierce westman...@gmail.com wrote: Is there a process for installing the ILBC codec under this environment, or will I have to un-install the RPMs and build Asterisk from source? You can write a short makefile for just codec_ilbc module, build it and install it on your running asterisk system. You will have to install the asterisk18-devel package and get the asterisk source code either from a tar or from the srpm. If you are familiar with the basics of writing makefiles its pretty trivial to write one that builds codec_ilbc, I have done this in numerous systems that use the digium rpms and it works flawlessly. This method can also be used to build other modules that are missing from the digium rpms. --- Lefteris Zafiris -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Hi Jonathan, Any clue with 1.6.2.19.*1 *might be released? Regards, On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Can you please point me to the patch that you just made? Thanks On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote: We worked on this bug today and are expecting to release packages with the fix soon, possibly tomorrow (Aug 2). The issue arose from a change in features reload which was back-ported to 1.6.2 and was committed without enough testing to observe the intermittent crash behavior. Thanks for your patience, Jonathan R. Rose - Original Message - From: Vahan Yerkanian va...@arminco.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 31, 2011 3:03:07 AM Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs? On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- 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] use ILBC installed from asterisk yum repositories
On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris zaf@gmail.com wrote: You can write a short makefile for just codec_ilbc module, build it and install it on your running asterisk system. You will have to install the asterisk18-devel package and get the asterisk source code either from a tar or from the srpm. If you are familiar with the basics of writing makefiles its pretty trivial to write one that builds codec_ilbc, I have done this in numerous systems that use the digium rpms and it works flawlessly. This method can also be used to build other modules that are missing from the digium rpms. Thanks for the pointer. I think I'll give this method a try. I'll see if I can figure out how to write the makefile. -- _ -- 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] use ILBC installed from asterisk yum repositories
There is much more to installing and configuring OOH323 as it's not easy breezy install. I think a professional developer help would be more appropriate than users patching. Just my thought.plus it adds a great deal of functionality to Asterisk to allow for all add-ons to be install via RPMS or at least the ones related to codec and protocols. On Tue, Aug 2, 2011 at 6:33 PM, Bob Pierce westman...@gmail.com wrote: On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris zaf@gmail.com wrote: You can write a short makefile for just codec_ilbc module, build it and install it on your running asterisk system. You will have to install the asterisk18-devel package and get the asterisk source code either from a tar or from the srpm. If you are familiar with the basics of writing makefiles its pretty trivial to write one that builds codec_ilbc, I have done this in numerous systems that use the digium rpms and it works flawlessly. This method can also be used to build other modules that are missing from the digium rpms. Thanks for the pointer. I think I'll give this method a try. I'll see if I can figure out how to write the makefile. -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Can you please point me to the patch that you just made? The patch is committed to v1.6.2 SVN branch. Patch for v1.6.2 only. r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines Asterisk 18103 - Fix reload crash caused by destroying default parking lot Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ Also -r330505 to fix a ref leak with the above patch. 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] TE410P hardware problems
TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and restarting dahdi with no luck. The other two cards are working fine. I put a loop plug the ports and same problem. dahdi_scan [9] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 1 name=TE4/2/1 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=193 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [10] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 2 name=TE4/2/2 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=217 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [11] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 3 name=TE4/2/3 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=241 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [12] active=yes alarms=RED/LFA/LMFA description=T4XXP (PCI) Card 2 Span 4 name=TE4/2/4 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=265 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF lspci 09:04.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) 0a:02.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) 0a:03.0 Communication controller: Digium, Inc. Wildcard TE410P/TE412P quad-span T1/E1/J1 card 3.3V (rev 02) Thanks Dave -- _ -- 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] Outgoing call issue in D-link DPH-80 ip phones
Hi All, Along with my asterisks server, all incoming calls to my D-link DPH-80 ip phones are are working fine while calling from soft phones with good voice clarity. But not able to make outgoing calls from the same D-link DPH-80 ip phones to either soft phone or IP phone. What would be the reason ? Any suggestions please Regards, Michael.k -- _ -- 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] snom and srtp
Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). -snip-- == Using SIP RTP CoS mark 5 -- Executing [1@default-outbound08:1] Dial(SIP/10002-0012, SIP/1,30) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1 -- SIP/1-0013 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [1@default-outbound08:2] VoiceMail(SIP/10002-0012, 1,uj) in new stack [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure -- SIP/10002-0012 Playing 'vm-theperson.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/1.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') sage*CLI Disconnected from Asterisk server [root@sage asterisk]# ---snip--- The interesting thing here is the call fails at this point and for some reason the cli disconnects when the call fails. Here is a call to a mobile which connects but the call dies in about 4 seconds --snip == Using SIP RTP CoS mark 5 -- Executing [0429835743@default-outbound08:1] Dial(SIP/10002-, SIP/private-sip/0429835743) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/private-sip/0429835743 -- SIP/private-sip-0001 is ringing -- SIP/private-sip-0001 answered SIP/10002- [Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure [Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure sage*CLI Disconnected from Asterisk server --snip I have done heaps of reading on SRTP unprotect error but cant really work it out from that. Q. should I try the patch mentioned below and forget about snoms doing 80 bit incription or should I persevere with making this work? thanks James ---snip--- Patch SRTP for 32bit SRTP have a cryptographic hash to check the integrity of the encrypted packets. It support two hash size: ● 32bit ● 80bit In order to properly fine tune SRTP for mobile networks and to have compatibility with PrivateGSM Enterprise we must use SRTP with hash at 32bit (HMAC_SHA1_32). Asterisk 1.8 by default does not announce in SDP both 32bit and 80bit, but only the 80bit version even if both are supported. This very small 1 line patch make Asterisk by default work with SRTP hash at 32bit . Download the patch for HMAC_SHA1_32 RTP crypto offer 48. wget http://sourceforge.net/projects/Asterisk-amr/files/1.8.0-rc2_crypto_offer.diff/download Apply the patch 49. cd Asterisk-1.8.0/ patch -p2 ../1.8.0-rc2_crypto_offer.diff Go to Asterisk-1.8.0/ folder50. cd .. Recompile Asterisk , 51. make ; make instal snip-- -- _ -- 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] TE410P hardware problems
On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote: TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and restarting dahdi with no luck. The other two cards are working fine. I put a loop plug the ports and same problem. Any strange output in dmesg? Best guess is either the card has failed or it has started to unseat from it's PCI slot. Normally when it starts to unseat you'll see Version Synchronization Errors in dmesg when trying to load the card. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] TE410P hardware problems
Hi, We opened the server an checked that the cards were seated correctly and they are. I will have the tech completely remove them tomorrow and try again. I will post the results. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, August 02, 2011 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote: TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and restarting dahdi with no luck. The other two cards are working fine. I put a loop plug the ports and same problem. Any strange output in dmesg? Best guess is either the card has failed or it has started to unseat from it's PCI slot. Normally when it starts to unseat you'll see Version Synchronization Errors in dmesg when trying to load the card. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] TE410P hardware problems
dmesg: wct4xxp :0a:03.0: SPAN 9: Primary Sync Source wct4xxp :0a:03.0: Span 2 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 10: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 wct4xxp :0a:03.0: Span 3 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 11: Primary Sync Source wct4xxp :0a:03.0: Span 4 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 12: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 system.conf span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,1,0,esf,b8zs bchan=26-48 mtp2=25 span=3,1,0,esf,b8zs bchan=49-72 span=4,1,0,esf,b8zs bchan=73-96 span=5,1,0,esf,b8zs bchan=97-120 span=6,1,0,esf,b8zs bchan=121-144 span=7,1,0,esf,b8zs bchan=145-168 span=8,1,0,esf,b8zs bchan=169-192 span=9,1,0,esf,b8zs bchan=193-216 span=10,1,0,esf,b8zs bchan=217-240 span=11,1,0,esf,b8zs bchan=241-264 span=12,1,0,esf,b8zs bchan=265-288 Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, August 02, 2011 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote: TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and restarting dahdi with no luck. The other two cards are working fine. I put a loop plug the ports and same problem. Any strange output in dmesg? Best guess is either the card has failed or it has started to unseat from it's PCI slot. Normally when it starts to unseat you'll see Version Synchronization Errors in dmesg when trying to load the card. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] TE410P hardware problems
If it doesn't go green when you put a hard loopback on the port, then contact Digium support. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave George Sent: Tuesday, August 02, 2011 10:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] TE410P hardware problems dmesg: wct4xxp :0a:03.0: SPAN 9: Primary Sync Source wct4xxp :0a:03.0: Span 2 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 10: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 wct4xxp :0a:03.0: Span 3 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 11: Primary Sync Source wct4xxp :0a:03.0: Span 4 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 12: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
On Tue, Aug 02, 2011 at 03:40:17PM -0500, Warren Selby wrote: Install OpenSSL-devel (or whatever the equivalent ubuntu package is called) libssl-dev and then recompile / reinstall and test it again. -- 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] CHANSPY
Hi, I am using Asterisk1.4, in need to configure barge of all SIP Id's. SIP Id's start from 900 to 999, configuration for barge using ChanSpy application. in extensions.conf exten = 81,1,ChanSpy(SIP) exten = 81,2,Hangup * for next barge But problem is at whenever 938 is comming at press * its going blank. every time its going blank at 938. how can I resolve please help. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
