Re: [asterisk-users] How to enable IM over the asterisk server
Hi, I think so yes unless somebody else can provide a better solution. (Perhaps I'm doing it wrong ;-) ) We have 2 asterisk servers (Xivo distribution based on Debian) whom work in Active/Passive cluster mode. Then we have a third server which is the OpenFire server (based on Ubuntu 14) So yes, we have 2 different servers for that. nb. The 3CX version of Asterisk (Windows Version) has a baked-in IM server so then you only need one. But it does require you to buy a license which I didn't want to because Asterisk is opensource. //Kristof Thyda ENG ength...@gmail.com 8/07/2015 10:24 I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof Thyda ENG ength...@gmail.com 7/07/2015 11:28 Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof Thyda ENG ength...@gmail.com 7/07/2015 6:07 I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by Cisa Antispam Service, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users
Re: [asterisk-users] How to enable IM over the asterisk server
You can have the openfire server installed on the same server as asterisk without any issue, just size your server appropriately. Just keep in mind they are different services. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote: I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof Thyda ENG ength...@gmail.com 7/07/2015 11:28 Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof Thyda ENG ength...@gmail.com 7/07/2015 6:07 I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by *Cisa Antispam Service*, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or
Re: [asterisk-users] How to enable IM over the asterisk server
I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof Thyda ENG ength...@gmail.com 7/07/2015 11:28 Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof Thyda ENG ength...@gmail.com 7/07/2015 6:07 I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by *Cisa Antispam Service*, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this
Re: [asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user
On 6 July 2015 at 15:27, Motty Cruz motty.c...@gmail.com wrote: Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks, You could use fail2ban for this by adding your own filter string specific for that user. It would have the advantage of blocking further calls as well as alerting you by email. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF issue
Indeed, thanks. I'll let you know how it goes. Thanks, Jamie -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 22:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DTMF issue You probably have to reload asrerisk after making the change. Thomas M. Peters | Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Jamie Rees jr...@gmlnt.com 7/7/2015 3:53 PM Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm not too confident it has. Thanks, Jamie -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 20:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DTMF issue In my humble opinion, adjusting this setting will (for you) do nothing, since you don't use the dahdi channels for transport. See this discussion, which I found after I posted my first response: http://www.voip-info.org/wiki/view/Asterisk+DTMF Particularly this sentence: Note: Asterisk 1.4 now also has the relaxdtmf= setting available in sip.conf. The big question for you is going to be, does your system need to recognize inbound DTMF tones, and if so, will setting relaxdtmf=NO cause problems doing that? Thomas M. Peters | Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Jamie Rees jr...@gmlnt.com 7/7/2015 2:03 PM Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer used. My dahdi-channels.conf file looks stock: ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 group=0,12 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 32-46,48-62 context = default group = 63 Thanks again, Jamie -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 19:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF issue It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile phones but it happens at random on many external calls. If this happens to you, especially on voice peaks (when the outside party said a particularly loud syllable) then you probably have DTMF talk-off. I think it's caused by an audio tone mistakenly being interpreted at a broken DTMF tone and getting regenerated by your T1 or POTS card, or Asterisk itself. We use a Digium T1 card and dahdi. We had reduced talk-off noticeably by using ... relaxdtmf=no ...in /etc/asterisk/dahdi-channels.conf (this is a per-channel setting) Problem with that it that our autoattendant wasn't recognizing DTMF tone from callers very well. They would dial 4 digits and in my logs, I'd see one or two, maybe three. The autoattendant would tell them they had dialed an invalid extension. So we had to go back to relaxdtmf=yes on the dahdi channels in question. So problem_solved=no. -T Thomas M. Peters | Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Jamie Rees jr...@gmlnt.com 7/6/2015 4:53 PM Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets. I have enabled DTMF logging and spoken to the SIP provider, but they couldn't really help much. I presume the issue is local to our phone system but other than the logs below, have nothing to go on: [2015-06-10 09:32:26] DTMF[3280][C-c5a1] channel.c: DTMF begin '2' received on SIP/sip-out-00021c6d [2015-06-10
[asterisk-users] תשובה: Asterisk how to setup alarm too many outgoing calls from same user
You could use the group functionCreate the group by extension and check how many calls are in the groupIf it's more than you allow then have it send a emailמאת: Ishfaq Malikנשלח: יום רביעי, 8 ביולי 2015 12:22אל: Asterisk Users Mailing List - Non-Commercial Discussionהשב ל: Asterisk Users Mailing List - Non-Commercial Discussionנושא: Re: [asterisk-users] Asterisk how to setup alarm too many outgoing calls from same userOn 6 July 2015 at 15:27, Motty Cruz motty.c...@gmail.com wrote:Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks, You could use fail2ban for this by adding your own filter string specific for that user. It would have the advantage of blocking further calls as well as alerting you by email.RegardsIsh-- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
Hi list, we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual running version. Patch failed with zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 ../asterisk-11.18.0-patch can't find file to patch at input line 5 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff --git a/.version b/.version |index c5df2aa..150754a 100644 |--- a/.version |+++ b/.version -- File to patch: It seems that patch file for 11.18.0 are completely different from previous one. Patch for 11.17.0 looked like (first 3 lines) --- asterisk-11.16.0-summary.html (.../11.16.0) (revision 433916) +++ asterisk-11.16.0-summary.html (.../11.17.0) (revision 433916) @@ -1,307 +0,0 @@ which is different from 11.18.0 diff --git a/.version b/.version index c5df2aa..150754a 100644 --- a/.version +++ b/.version @@ -1 +1 @@ OS is Debian Wheezy 7.8 What are we doing wrong ? Thanks for any hint -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I use ARI to update the builtin database, without executing the dial plan?
Rodrigo Pimenta Carvalho wrote: Hi. In my dial plan I can use the following commands to access and handle data from the builtin database. DB DB_DELETE DB_EXISTS DB_KEYS Equivalents exist for these in the Asterisk Manager Interfaces as actions. An example being DBGet[1] but others also exist. They require no channel to operate on. From the scope of ARI though you can use the /asterisk/variable resource to GET and POST global variables and dialplan functions[2]. This also requires no channel to operate on. I haven't personally tried it for this use case though. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DBGet [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?
Richard Kenner wrote:
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer
= 100021, format = { id = AST_FORMAT_SLINEAR16, fattr = {format_attr
= { 0repeats 64 times}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64, src
= 0x51623b0 func_jitterbuffer interpolation, data = {ptr = 0x0,
uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
so datalen is 0 and samples nonzero. ast_frame_adjust_volume,
however, iterates over samples, not datalen. Is that correct?
What does it mean to have a packet with a zero datalen anyway?
This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] How may SIP 183 messages a caller receives when many callee rings?
Rodrigo Pimenta Carvalho wrote: Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten = 2005,1,Dial(SIP/2000SIP/2001SIP/2002, 30) ' All softphones (2000, 2001 and 2002) will ring. These are proprietary softphones and all of then will reply with SIP 183 message. SIP 183 will contain SDP with media information. The question is: Will the caller receive SIP 183 from each callee? That is, will it receive 3 SIP 183 messages? It is important to the caller receives a SIP 183 message from each callee, because this caller needs to send early media (video) to every callee. Or, will Asterisk send just one message SIP 183 to the caller, with some kind of generic SDP message? Asterisk isn't a proxy, so it won't forward all 3 and it won't forward media from all 3. Right now the Dial application is simple and just doesn't forward media in this scenario. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net wrote: Hi list, we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual running version. Patch failed with zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 ../asterisk-11.18.0-patch can't find file to patch at input line 5 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff --git a/.version b/.version |index c5df2aa..150754a 100644 |--- a/.version |+++ b/.version -- File to patch: It seems that patch file for 11.18.0 are completely different from previous one. Patch for 11.17.0 looked like (first 3 lines) --- asterisk-11.16.0-summary.html (.../11.16.0) (revision 433916) +++ asterisk-11.16.0-summary.html (.../11.17.0) (revision 433916) @@ -1,307 +0,0 @@ which is different from 11.18.0 diff --git a/.version b/.version index c5df2aa..150754a 100644 --- a/.version +++ b/.version @@ -1 +1 @@ OS is Debian Wheezy 7.8 What are we doing wrong ? The two patch files were created by different version control systems. One was created by git the other created by subversion. For the git patch you would need to use -p1 for the subversion patch you would need to use -p0. The patch program gave you this hint when it failed to apply the patch: Perhaps you used the wrong -p or --strip option?. 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] How to handle multiple lines call
Hi,
I am new to asterisk, I have set up the asterisk server and successfully I
could make the dialplan between 2 SIPs but when there are more than two
sips calling each other, my dialplan seems doing the wrong routing to the
sip. Do i need to config anything additionally to asterisk to handle this?
Example:
we have 6 sips
-sip 1 is calling to sip 2
-sip 3 is calling to sip 4
-sip 5 is calling to sip 6
Here is my configuration,
exten = _.,1,Dial(SIP/${EXTEN})
I am waiting for your reply, Thank.
Thyda
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Re: [asterisk-users] How to handle multiple lines call
On Wednesday 08 Jul 2015, Thyda ENG wrote:
Hi,
I am new to asterisk, I have set up the asterisk server and successfully I
could make the dialplan between 2 SIPs but when there are more than two
sips calling each other, my dialplan seems doing the wrong routing to the
sip. Do i need to config anything additionally to asterisk to handle this?
Example:
we have 6 sips
-sip 1 is calling to sip 2
-sip 3 is calling to sip 4
-sip 5 is calling to sip 6
Here is my configuration,
exten = _.,1,Dial(SIP/${EXTEN})
That looks about right, but it's not quite enough information.
Your sip.conf should have the SIP device descriptions in sections labelled [1]
, [2] , [3] , [4] , [5] and [6] , since the bit between the square brackets is
what gets matched against the SIP/ bit in the Dial() statement.
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Re: [asterisk-users] How to enable IM over the asterisk server
Yes, I have though of setting them up on the same server(openfire, and asterisk) and the problem come in mind that how can register the user to openfire automatically when I register the user SIP on the asterisk server ? Do you have any idea? I am waiting for your reply. Thank, Thyda On Wed, Jul 8, 2015 at 6:55 PM, James Cass jcas...@gmail.com wrote: You can have the openfire server installed on the same server as asterisk without any issue, just size your server appropriately. Just keep in mind they are different services. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote: I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof Thyda ENG ength...@gmail.com 7/07/2015 11:28 Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof Thyda ENG ength...@gmail.com 7/07/2015 6:07 I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by *Cisa Antispam Service*, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] How to enable IM over the asterisk server
The asterisk plugin for openfire would be what I would think would do that, but as another person posted, it's very deprecated, so I'm not sure how well it would work. I've never used it personally. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 11:45 AM, Thyda ENG ength...@gmail.com wrote: Yes, I have though of setting them up on the same server(openfire, and asterisk) and the problem come in mind that how can register the user to openfire automatically when I register the user SIP on the asterisk server ? Do you have any idea? I am waiting for your reply. Thank, Thyda On Wed, Jul 8, 2015 at 6:55 PM, James Cass jcas...@gmail.com wrote: You can have the openfire server installed on the same server as asterisk without any issue, just size your server appropriately. Just keep in mind they are different services. James Cass http://goog_987864563 jcas...@gmail.com On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote: I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof Thyda ENG ength...@gmail.com 7/07/2015 11:28 Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland kvandenouwel...@vangenechten.com wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof Thyda ENG ength...@gmail.com 7/07/2015 6:07 I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by *Cisa Antispam Service*, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of
Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?
This is an interpolated frame from func_jitterbuffer. It's part of packet loss concealment. What scenario exposed this? We were testing for clipping by doing Set(VOLUME(RX)=100) but we were connecting to a ConfBridge that had a jitterbuffer. This occurred when the phone (SIP) hung up. -- _ -- 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] Call Return
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?
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[asterisk-users] RES: How many SIP 183 messages a caller receives when many callee rings?
Hi Joshua Colp. Thank you very much for alerting me about the impossibility of forwarding the SIP 183 messages from callees to caller, via Asterisk, when more than 1 callee ring at same time. In my project the caller software (a proprietary softphone) needs to know some information about the callees, while they are still all ringing. Such information will be used to create early media (only video) from caller to all callees. For example, the caller softphone should receive the IPs and ports where each callee will listen to video data. The caller softphone will use RTSP to create such early media. That is why I was investigating an way of passing SIP 183 messages from callees to the caller. However, as you told me about such impossibility, now I have to discover a way of collecting such callees' media information and deliver it to the proprietary caller software. So, I ask you: 1 - Is there a way of collecting information from SIP messages that arrives in Asterisk, in dial plan (by means of application or functions)? If yes, I could pass it to a external software. 2- Is there a way of handling SIP 183 or SIP 180 messages in dial plan and forward such messages to another destiny, as in a proxy? 3 - Should I use Asterisk REST Interface to collect information from SIP messages that pass in the current channel of a call, whether I need collect it and pass to a proprietary software? I was reading about ARI today. 4 - By the way, can an external application, using ARI, send requests to the Asterisk, even when such application is not invoked by a dial plan? That is, can an external application decide by itself to contact a Asterisk REST interface? Any hint about early media (video) with asterisk will be very helpful to me, as I'm completely beginner in this field. Thanks a lot! RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Joshua Colp [jc...@digium.com] Enviado: quarta-feira, 8 de julho de 2015 11:53 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] How may SIP 183 messages a caller receives when many callee rings? Rodrigo Pimenta Carvalho wrote: Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten = 2005,1,Dial(SIP/2000SIP/2001SIP/2002, 30) ' All softphones (2000, 2001 and 2002) will ring. These are proprietary softphones and all of then will reply with SIP 183 message. SIP 183 will contain SDP with media information. The question is: Will the caller receive SIP 183 from each callee? That is, will it receive 3 SIP 183 messages? It is important to the caller receives a SIP 183 message from each callee, because this caller needs to send early media (video) to every callee. Or, will Asterisk send just one message SIP 183 to the caller, with some kind of generic SDP message? Asterisk isn't a proxy, so it won't forward all 3 and it won't forward media from all 3. Right now the Dial application is simple and just doesn't forward media in this scenario. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tls on asterisk 13
2015-07-08 13:11 GMT-06:00 Joshua Colp jc...@digium.com: You probably want to add rewrite_contact=yes to your endpoint. This will cause it to reuse the existing connection established from the phone. Generally the port provided by the phone is not reachable. Hi Joshua , I add the option you recommended but still can not connect, the strange thing is that I get another message always using TLS transport [Jul 8 14:28:45] NOTICE[2498]: res_pjsip/pjsip_distributor.c:256 log_unidentified_request: Request from 'X00X sip:X00X@172.16.8.55' failed for '172.16.8.179:5065' (callid: 5ece51c0-9ed5173a@172.16.8.179) - No matching endpoint found --- Transmitting SIP response (479 bytes) to TLS:172.16.8.179:5065 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 172.16.8.179:5065;rport=5065;received=172.16.8.179;branch=z9hG4bK-27b9198a Call-ID: 5ece51c0-9ed5173a@172.16.8.179 From: X00X sip:X00X@172.16.8.55;tag=ff2e31b0cc3d380ao3 To: sip:172.16.8.55;tag=z9hG4bK-27b9198a CSeq: 54 NOTIFY WWW-Authenticate: Digest realm=asterisk,nonce=1436387325/20cc7b903ffd92277b22c633e27854de,opaque=5b36911758ac6b0e,algorithm=md5,qop=auth Server: Asterisk PBX 13.4.0 Content-Length: 0 regardss -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP, T.38 fax gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, I'm trying to receive fax from PSTN, with the following setup: Fax machine --- PSTN --- *11 --- *13 --- IAXmodem + Hylafax Fax machine is connected to the PSTN, call arrives via ISDN on Asterisk 11.16.0 used as gateway, chan_sip relays the call to Asterisk 13.4.0 receiving via chan_pjsip. I'm trying to have T.38 working between the 2 Asterisk servers: I've done that with success with both Asterisk running 11, but I can't make it work with Asterisk 13. I think the configuration is correct, as the traces below show that T.38 is negotiated correctly, but there is always only one UDPTL packet transmitted from Asterisk-13 to Asterisk-11: wireshark shows UDPTLPacket t30ind: no-signal Is it a bug in chan_pjsip, or did I miss something? Here is the SIP trace on the gateway: == Primary D-Channel on span 2 up -- Accepting call from '40483527' to '1041' on channel 0/1, span 2 -- Executing [1041@entrant_rnis:1] NoOp(DAHDI/i2/40483527-18e, Appel entrant sur ligne RNIS) in new stack -- Executing [1041@entrant_rnis:2] Set(DAHDI/i2/40483527-18e, FAXOPT(gateway)=yes) in new stack -- Executing [1041@entrant_rnis:3] Dial(DAHDI/i2/40483527-18e, SIP/tiare/1041) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 7740 Adding codec 14 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.200:5060: INVITE sip:1041@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK445324c9 Max-Forwards: 70 From: sip:40483527@192.168.0.10;tag=as40626b30 To: sip:1041@192.168.0.200 Contact: sip:40483527@192.168.0.10:5060 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.16.0 Date: Thu, 09 Jul 2015 05:02:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: 40483527 sip:40483527@192.168.0.10 Content-Type: application/sdp Content-Length: 233 v=0 o=root 687045483 687045483 IN IP4 192.168.0.10 s=Asterisk PBX 11.16.0 c=IN IP4 192.168.0.10 t=0 0 m=audio 7740 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv - --- -- Called SIP/tiare/1041 --- SIP read from UDP:192.168.0.200:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c 9 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 From: sip:40483527@192.168.0.10;tag=as40626b30 To: sip:1041@192.168.0.200 CSeq: 102 INVITE Server: Asterisk GPL PBX Content-Length: 0 - - --- (8 headers 0 lines) --- --- SIP read from UDP:192.168.0.200:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c 9 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 From: sip:40483527@192.168.0.10;tag=as40626b30 To: sip:1041@192.168.0.200;tag=c0f238c0-f4de-4c5b-b8b4-3c8570b40620 CSeq: 102 INVITE Server: Asterisk GPL PBX Contact: sip:192.168.0.200:5060 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER Content-Length: 0 - - --- (10 headers 0 lines) --- list_route: hop: sip:192.168.0.200:5060 -- SIP/tiare-0165 is ringing --- SIP read from UDP:192.168.0.200:5060 --- OPTIONS sip:ti...@gw.sysnux.pf:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7 ff33b From: sip:db25221c-c317-4185-9c7d-050cd9377012@192.168.0.200;tag=f6a90675-24 14-4365-a46e-1678844bee7d To: sip:ti...@gw.sysnux.pf Contact: sip:db25221c-c317-4185-9c7d-050cd9377012@192.168.0.200:5060 Call-ID: b5dab0f5-d07b-461b-aa16-a5a9aa93369a CSeq: 22129 OPTIONS Max-Forwards: 70 User-Agent: Asterisk GPL PBX Content-Length: 0 - - --- (10 headers 0 lines) --- Sending to 192.168.0.200:5060 (no NAT) Looking for tiare in none (domain gw.sysnux.pf) --- Transmitting (no NAT) to 192.168.0.200:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7ff33b; received=192.168.0.200;rport=5060 From: sip:db25221c-c317-4185-9c7d-050cd9377012@192.168.0.200;tag=f6a90675-24 14-4365-a46e-1678844bee7d To: sip:ti...@gw.sysnux.pf;tag=as0e01251c Call-ID: b5dab0f5-d07b-461b-aa16-a5a9aa93369a CSeq: 22129 OPTIONS Server: Asterisk PBX 11.16.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog 'b5dab0f5-d07b-461b-aa16-a5a9aa93369a' in 32000 ms (Method: OPTIONS) --- SIP read from UDP:192.168.0.200:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c 9 Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060 From:
Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
Le 08/07/2015 17:36, Richard Mudgett a écrit : On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Hi list, we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual running version. Patch failed with zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 ../asterisk-11.18.0-patch can't find file to patch at input line 5 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff --git a/.version b/.version |index c5df2aa..150754a 100644 |--- a/.version |+++ b/.version -- File to patch: [SNIP] The two patch files were created by different version control systems. One was created by git the other created by subversion. For the git patch you would need to use -p1 for the subversion patch you would need to use -p0. The patch program gave you this hint when it failed to apply the patch: Perhaps you used the wrong -p or --strip option?. We already tried with no luck: zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1 ../asterisk-11.18.0-patch patching file .version Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file .version.rej patching file ChangeLog Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file ChangeLog.rej The next patch would delete the file asterisk-11.18.0-rc1-summary.html, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored The next patch would delete the file asterisk-11.18.0-rc1-summary.txt, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored patching file asterisk-11.18.0-summary.html patching file asterisk-11.18.0-summary.txt As you can see, patch is against -rc1 not 11.17.0 ... Thanks for your support. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tls on asterisk 13
2015-07-08 13:09 GMT-06:00 Ryan, Travis ry...@oscarwinski.com: Asterisk13 can do native tls with each phone? Nice. any example? rickygm http://gnuforever.homelinux.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
Re: [asterisk-users] How to handle multiple lines call
Hi Thyda
When you set exten = _.,1,Dial(SIP/${EXTEN}) Asterisk assume _., an match
everything on your dialplan including special extensions as i, h, etc.,
these will cause problems onto your system.
If you need to match one or more digits you can use _x and _x.
_x it mean only one pattern digit form 0 to 9
_x. any pattern digit from 0 to 9 and dot it mean remnant digits could be 2 or
3, 4 ... etc., so what ever you dial on sip it will be valid.
Mc GRATH Ricardo
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Re: [asterisk-users] tls on asterisk 13
ricky gutierrez wrote: Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) [Jul 8 11:09:46] ERROR[14733]: pjsip:0?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:46] WARNING[14733]: pjsip:0?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=31917 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) someone has had good results with tls my config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=@default auth= aors= media_encryption=sdes dtmfmode=rfc4733 You probably want to add rewrite_contact=yes to your endpoint. This will cause it to reuse the existing connection established from the phone. Generally the port provided by the phone is not reachable. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tls on asterisk 13
Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 ?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) [Jul 8 11:09:46] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:46] WARNING[14733]: pjsip:0 ?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=31917 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) someone has had good results with tls my config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=@default auth= aors= media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.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
Re: [asterisk-users] tls on asterisk 13
Asterisk13 can do native tls with each phone? Nice. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ricky gutierrez Sent: Wednesday, July 08, 2015 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] tls on asterisk 13 Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed to make it work, all my terminals spa Cisco 5XX look my cli [Jul 8 11:09:16] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:16] WARNING[14733]: pjsip:0 ?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=48024 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) [Jul 8 11:09:46] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS connect() error: Connection refused [code=120111] [Jul 8 11:09:46] WARNING[14733]: pjsip:0 ?: tsx0x7f53a8008 Failed to send Request msg OPTIONS/cseq=31917 (tdta0x7f53c000dcb0)! err=120111 (Connection refused) someone has had good results with tls my config [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=@default auth= aors= media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.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 -- _ -- 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
