Re: [asterisk-users] Inexpensive device for bandwidth management
Mike, This firmaware works on Buffalo, linksys and some asus routers. Linksys did release the wrt54gL because of the demand to have a router with Linux. In fact, the L means Linux and this router is still in production, easy to find (in Europe anyway) and very very cheap. DD-Wrt also runs on it and some other Linux firmware with QOS. The reason I recommand Tomato is that Tomato is very easy to configure for end users. DD-wrt is more complete but also more complex. Mike a crit: I just reread my question and realized I might not have been clear enough. What I meant is that it only seems to works on older Linksys hardware revisions. How do I make sure I can get those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Sunday, April 05, 2009 15:30 To: oliv...@hh174.be; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Actually that was my original thought. BUTaccording to what I read on their FAQ, the hardware that can be used is rather limited. How do I secure a reliable supply of those? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hh174 Sent: Sunday, April 05, 2009 14:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management Linksys (cisco)WRT54GL and the tomato firmware. 5 minutes setup Olivier Mike a crit: Thanksthe thing is I need many device (one for each of my hosted customers) and I'd like this process to be as easy for non-techies as possible, because some of those are technologically-challenged, and need to install the box by themselves or with the help of an IT person that only knows how to install a run of the mill router. So an out-of-the-box thing would be better, but I was recommende the pfsense before and will take a look at it. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Sunday, April 05, 2009 13:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive device for bandwidth management The following two links deal with the same familly of boxes. Generally it's $20 for a case, $20 for a powersupply, but you've probably got an old one that will work. and almost all of their boards are under $200, except for the ones with lots of gigabit interfaces. Many are under $100. http://www.mikrotik.com/ http://routerboard.com/ On Sun, Apr 5, 2009 at 11:07 AM, Mike l...@virtutel.ca wrote: Hi, I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable enough for my taste), both from D-link. What else is out there and allows me to do upstream QoS on cable/DSL links? Both D-Link routers were under 200$ (99$ and 159$ respectively) and were perfect price-wise for my target customers. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Einhorn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global h exten
I had a patch created for 1.4.X for this. http://bugs.digium.com/bug_view_page.php?bug_id=14159 - Original Message - From: Gabriel Ortiz Lour To: asterisk-users@lists.digium.com Sent: Wednesday, March 18, 2009 8:23 PM Subject: [asterisk-users] Global h exten Hi all, Is there something like a global h exten, that gets called on every hang up, no matter what exten? Thanks, Gabriel -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global h exten
I had a patch created for 1.4.X for this. http://bugs.digium.com/bug_view_page.php?bug_id=14159 - Original Message - From: Gabriel Ortiz Lour To: asterisk-users@lists.digium.com Sent: Wednesday, March 18, 2009 8:23 PM Subject: [asterisk-users] Global h exten Hi all, Is there something like a global h exten, that gets called on every hang up, no matter what exten? Thanks, Gabriel -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] async agi question
Hi, I was asked for the patch and I sent it. Does anybody have any news about this subject? I'm willing to try a fix for 1.4 but I'd need any guidelines to do it. Thanks in advanced Jose 2009/4/2 Moises Silva moises.si...@gmail.com Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something, do you still have the patch around? Moy On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote: Hi Henrik, I would like to do the same thing you are doing here. I want to implement an external queue functionality so I need to stop a play file launched previously with an async agi command on caller's channel, sending the call to agent's extension. I'm redirecting caller's channel with REDIRECT while playing is taking place but I'm always getting a hang up on caller's channel. I'm using: asterisk-1.4.18 asterisk-addons-1.4.7 async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4) Both caller and agent are using 501 and 500 extensions and the async agi loop is waiting on 800, for example. The caller is dialing 800 where a play file is commanded through and async agi stream file command by the application. The relevant part of extensions.conf follows: exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN}); exten = _5.,n,Wait(1); exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto); exten = _5.,n,Hangup(); exten = _8.,1,Noop(every thing starting 8 ${EXTEN}); exten = _8.,n,AGI(agi:async); exten = _8.,n,Hangup(); And the redirect command the application is sending to is: Action: Redirect Channel: SIP/501-081f0730 Exten: 500 Context: sip_sercom Priority: 1 Therefore, Henrik, could you show me your related dial plan and the redirect command you are sending? I wasn't able to see what I'm getting wrong. thanks in advanced Jose M Arias -- This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-topic: SIP DTMF most supported method
Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
Hi, we are using version 2.0.4 (vicidialnow distribution) now for some time in productino - working quit nice. Is there any upgrade instruction out there - or will a simple yum update do the job in the feature. PS: On the astguiclient site you have April 3, 2008 - Released version 2.0.5 - i think thats not correct ;-) regards, Wolfgang Am Freitag, den 03.04.2009, 10:30 -0400 schrieb Matt Florell: Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have added hundreds of new features including Asterisk phone, trunk and DID configuration through the VICIDIAL web interface. We have also tested the suite on Asterisk versions through 1.2.30.2 and 1.4.21.2. All client web-apps and administration pages are available in English, Spanish, Greek, German, Italian and French, with rough translations of Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice SIP 183 progessl
Dear Martin Can you inform me how to make the patch or from where I can get it otherwise if there is an application can generate it? Or if its relate to chan_sip.c ,please can you tell me which function to edit or lines to be added Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Sunday, April 05, 2009 5:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please Advice SIP 183 progessl Hi Khaled, app Dial clearly is coded to ignore the 180 Ringing being passed if you have 'm' option to Dial and you do. Try to take the 'm' out and see if 180 Ringing is passed to the A-leg. So if you want MOH and then when 180 Ringing comes to turn it off = you need a patch. Martin 2009/4/4 Khaled W. Chehab kche...@xplorium.com: 10x Martin , But B-Leg is sending 180 ringing Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card
On Mon, 6 Apr 2009, Tzafrir Cohen wrote: On Sun, Apr 05, 2009 at 11:35:18PM +0200, Puskás Zsolt wrote: On Sunday 05 April 2009 21.28.48 Gergo Csibra wrote: Saturday, April 4, 2009, 3:13:12 PM, Puskás wrote: Got it working with Asterisk 1.2 installed on the same PC as Asterisk 1.4 [ in different directorys and username of course :) ] . Using isdn4linux kernel module and Dial(Modem/ttyI0/1234567:${EXTEN}) command. Használj MISDN-t, és ne toppostolj. Use MISDN, and do not toppost! I won't toppost again but you should read my first e-mail again. I got a passive diva isdn card which are not supported: pc:~# mISDN scan 0 mISDN compatible device(s) found: chan_modem is deprecated. Any chance this works with chan_capi? Yes, but you need to use an older Version of Dialogic/Eicon driver package where support of these old cards was still part of. The Binary drivers Diva PCI cards as well as the Diva PRO supported full CAPI. Armin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4
Hi Philipp! On Sunday 05 April 2009, Philipp von Klitzing wrote: Take a look at these two links: Thanks for the links! So one option is to implement domain based authentication, which would be quite a bit of work. Another option which is quite popular is using an openSER (one of the two forks) in front of asterisk. Do you know if there is a way to keep sip device state in asterisk (for queues) working when using OpenSER as a sip proxy? Cheers, Florian -- DI Florian Hackenberger flor...@hackenberger.at www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
Implementing support for configuration of skills using an XML file would require rewriting one function. Adding the skill selections as an option of the queue would require a few lines of code. Apart from that your proposal pretty much matches my implementation. Cheers, Florian On Sunday 05 April 2009, nik600 wrote: Thanks, this is interesting. I'm still looking with a customer on a possible implementation of sbr, this is my proposal: Example of skill.conf [default] ; ; STATIC OR DYNAMIC DEFINITION ; ;skillpath=/etc/asterisk/skills.xml skillpath=http://x.x.x.x/skillgenerator.php ; STATIC DEFINITION [SIP/200] sbr_theme=,1 sbr_theme=,1 [SIP/201] sbr_theme=,1 sbr_theme=,1 * Example of XML file located in /etc/asterisk/skills.xml / or generated by http://x.x.x.x/skillgenerator.php skills member interface=default skill theme=z1/skill skill theme=y2/skill /member member interface=SIP/200 skill theme=z2/skill skill theme=y1/skill /member member interface=SIP/300 skill theme=y1/skill skill theme=x2/skill /member /skills * you can set some variables in the channel before to queue it: QUEUE_SBR_THEME_z QUEUE_SBR_THEME_y QUEUE_SBR_THEME_x you can also set in queues.conf the theme for each queue [queueA] sbr_theme=z sbr_theme=y sbr_theme=x On Sun, Apr 5, 2009 at 3:57 PM, Florian Hackenberger f.hackenber...@chello.at wrote: On Sunday 08 March 2009 17:11:33 nik600 wrote: Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? Hi! I have written a patch implementing skill based routing for asterisk 1.4.17 (can be ported to later versions quite easily). It works like this: You define a database table which stores the skills: columns: membername, skillname, skill_level You set the strategy to skill based and set a variable for each incoming call which specifies which skills to take into account, the weight of the skill and the minimum level (optional). When selecting agents to ring, asterisk picks the agents according to the highest value of weighted skills (skill level multiplied by skill weight for all skills taken into account for that particular call). If an agent does not satisfy the minimum, this agent does not ring at all. You can for example use the minimum to make sure only agents speaking a particular language get a call which requires that language. The implementation is finished and we are currently testing it. Unfortunately I'm quite busy at the moment and it may take about 2 months before I can take the time to release the code. Unless someone hires me as a consultant to work on it. Cheers, Florian -- DI Florian Hackenberger flor...@hackenberger.at www.hackenberger.at -- DI Florian Hackenberger flor...@hackenberger.at www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problem
Thanks for the tip, Harry. I will try that when I have exhausted all avenue. My problem is that if I upgrade to 1.4.24 and DAHDI, I'll break other stuffs. In my current set up, the PRI did work for a long period of time (7 hours) before going into this unreliable mode (up and down). I'm getting the telco technicians to check on this first because I believe the problem comes from their side. During the few hours when it is working, I am able to make and receive calls. So I don't think the issue lies with Asterisk. Regards, Steve Harry Vangberg wrote: I had the exact same problem and errors some time ago (search the archives for PRI dropping #2) using Asterisk 1.4.18, Zaptel and a Digium TE121. I tried all kind of things, had telco technicians come out and whatnot. The solution was two-folded - 1) I reinstalled my server, 2) I updated to Asterisk 1.4.24, replaced Zaptel with latest DAHDI. In the DAHDI case I even had to use latest Subversion revision due to some bug (but that was related to the TE121-cards I think). Since then I haven't had any issues at all, so consider updating Asterisk and Zaptel-DAHDI 2009/3/31 Steven J. Douglas stev...@moij.biz: Hi Brandon, When using the current straight cable, it sometimes worked i.e. I can make calls from the PSTN into the asterisk. Do you still think that I should try a crossover cable? Thanks. Regards, Steve. Brandon B. wrote: Try a T1 crossover cable: http://www.voip-info.org/wiki/view/crossover+T1+cable On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.biz mailto:stev...@moij.biz wrote: Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a straight cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same. /*etc/zaptel.conf* loadzone=sg defaultzone=sg # PRI Span span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 */etc/asterisk/zapata.conf* language=en progzone=sg musiconhold=default ; PRI Set Up context=inbound-pri1 switchtype=euroisdn signalling=pri_cpe pridialplan=national overlapdial=yes immediate=no faxdetect=both overlapdial=no usecallerid=yes usecallingpres=yes callerid=asreceived group=9 channel = 1-15 channel = 17-31 The following are the messages that keep repeating. == Primary D-Channel on span 1 down Mar 31 14:34:05 WARNING[2361]: chan_zap.c:2682 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 1 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 2 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 3 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 4 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 5 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 6 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 7 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 8 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 9 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 10 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 11 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 12 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 13 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 14 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 15 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 17 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 18 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 19 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 20 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel 21 Mar 31 14:34:07 NOTICE[2362]: chan_zap.c:6880 handle_init_event: Alarm cleared on channel
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
Mark Michelson wrote: Caution: One shortcoming of queue member penalties is that they are not taken into account if a queue member of a low penalty does not answer a call. Say for instance that the queue application determines that there are two members available to answer an incoming call. One member has penalty 1 and the other has penalty 2. If the member with penalty 1 does not answer the call, the queue application still considers that member to be available the next time that it tries to reach a member. The member with penalty 2 will only be tried if the queue application can determine *before the call is placed* that the member with penalty 1 is unavailable. Mark Michelson There is a patch http://bugs.digium.com/view.php?id=9165 which creates new strategy xrrmemory. It tries to call the member with higher penalty if queue member of a lower penalty does not answer the call. Unfortunately disclaimer was not submitted and this patch was suspended. We use this patch successfully in production and in fact this is the most popular strategy for many of our clients. It allows to create one queue and you can manage which members to call first, second and so on. For example, you have SIP/101,0 SIP/201,1 SIP/202,1 SIP/301,2 First queue will try to call SIP/101, then SIP/201, then SIP/202 (or SIP/202, SIP/201), and then SIP/301, after that it will call again SIP/101 and so on. You can also create simple linear strategy with dynamic members and they will be called in the order YOU set and not the order they are added to queue (current linear strategy). I know this can be done using QUEUE_MAX_PENALTY variable but you will need no enter queue and leave queue many times. This patch was written when the creator worked for our company so I can resubmit the code with license if it's interesting. Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fail to retrieve the calling party information
HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
On 4/6/09, Wolfgang Pichler wpich...@yosd.at wrote: Hi, we are using version 2.0.4 (vicidialnow distribution) now for some time in productino - working quit nice. Is there any upgrade instruction out there - or will a simple yum update do the job in the feature. PS: On the astguiclient site you have April 3, 2008 - Released version 2.0.5 - i think thats not correct ;-) Hello, There is actually an UPGRADE file right in the main directory of the release that you should read over. Since there are many database and dialplan changes since 2.0.4 a software-only upgrade would only get you part of the way. Thanks for the catch on the date, it has been fixed now. MATT--- Am Freitag, den 03.04.2009, 10:30 -0400 schrieb Matt Florell: Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have added hundreds of new features including Asterisk phone, trunk and DID configuration through the VICIDIAL web interface. We have also tested the suite on Asterisk versions through 1.2.30.2 and 1.4.21.2. All client web-apps and administration pages are available in English, Spanish, Greek, German, Italian and French, with rough translations of Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_queue.c: No one is answering queue
Hi all, Lastly we are getting several of the following errors: app_queue.c: No one is answering queue And when you isse a queue show XXX the status of the peers are reported as Invalid. We tried 1.4.23.1 and reverted back to 1.4.18.1 because it has showed good behaviour in the past but no luck, the queue randomly stops working due to the previous error. We have call-limit for each agent, which are static ones. We have checked that asterisk loads chan_sip before app_queue. Doing reload app_queue solves the issue and the agents reports the right status Can anyone explain under which circumstances the sip_devicestate http://www.asterisk.org/doxygen/1.4/chan__sip_8c.html#bd50965f28e901cb91713b8930eaa206 functions can return Invalid?? Is there any other place to look for the possible error? network connectivity? DNS error? Many thanks in advance, Samuel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mysql cache delay
Hi all, I use a mysql table for sip users and I fixed rtcachefriends param to yes in order to have a caching of this table. I would like to know how often does Asterisk check the mysql table to update its caching please. Regards, Cédric. -- Cédric Bonnet /FT/NCPI/DPS/CTR/CPM/VASF Tel. +33 (0) 1 55 88 36 60 cedric.bon...@orange-ftgroup.com * This message and any attachments (the message) are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Security
If that someone is between you and the other endpoint (like between you and the switch, or using port-mirroring on a router somewhere), then yes. The conversations can be recorded. In the US, the ability to be able to do this is required by law. You've little to worry about random hackers coming in off the Internet for this sort of thing. It's usually something to do with having physical access to the network in which you or the other end is connected. There's ARP poisoning and the like which could make this possible in a local network environment on either side, but for the most part, you'll know who's on your local net, and they likely have physical access to your phones as well. A listening device would be easier to plant in the mic pickup of your phone if they REALLY wanted to listen in on your calls. There are all sorts of levels one can to to find out what you're doing, and preventing against them can involve a great deal of creativity. That said, the answer is yes. You could use a VPN tunnel from one end to the other, and many people do just that to help ensure the privacy of their connections (both data and voice). N. Tom wrote: Since we are talking about security, if I am using * to talk to a cisco gateway via SIP, is there some sort of encryption you can use? Like a vpn tunnel? Can someone capture packets and re-assemble to make out a conversation? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Saturday, April 04, 2009 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Security Lets not be that paranoid. If you have these ports open to the internet then from time to time someone will check if your default unsecured context can dial out to PSTN... with sip.conf you can add allowguest=no With IAX2 there's no allowguest but I believe you have to have a guest username in iax.conf with no password to access the unsecured context. Martin On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese trees...@gmail.com wrote: Hi All, Coming in to day, the logs on the asterisk server showed several entries such as: [Apr 4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle_request_invite: Call from '' to extension '9810380487965419' rejected because extension not found. This has gotten me to thinking about security of this box. 1. Currently the box sits behind a firewall with iax and sip ports pointing to it for the ip phones that are offsite. There isn't any other access through the firewall to this box. 2. All devices have an extension assigned to them in sip.conf and extensions.conf. i.e. supra ATA, Grandstream GXP-2000 3. The box is fed via Les.net and Voicepluse. All other feeds are shutoff when not active. I'm looking for ideas to tighten up on the security so that this won't happen again. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.41/2040 - Release Date: 04/04/09 16:53:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Timer T309
Martin escreveu: What is the specification for T309 ? I'm too lazy to look it up. The default behaviour when the alarm of layer 1 (electrical T1/E1) is detected is to assume all calls dropped on both sides and that's what Asterisk does. The timer is simply deactivated since all the calls are supposed to drop. I believe that agrees with Q921/Q931 specs. Martin On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote: Hi everione, I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco link in this scenario: Telco Phone -- Telco --- Asterisk Sip Phone When i make a call from Telco Phone to Sip Phone, the call complete, but when i disconnect the link and reconnect in few seconds, the Asterisk clear call: [Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 1: Red Alarm [Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on 'DAHDI/1-1' [Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 2: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 2: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 3: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 3: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 4: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 4: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 5: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 5: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 6: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 6: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 7: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 7: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 8: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 8: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 9: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 9: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 10: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 10: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 11: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 11: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 12: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 12: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 13: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 13: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 14: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 14: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 15: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 15: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 17: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 17: Invalid argument [Apr 3 10:44:40]
[asterisk-users] 25-50-100fxs
Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Registration and INVITE question
I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mountain ahead of me!
Hello, I want to set up a Voip Farm (c) (tm) (patent pending) but don't know how to do it. Please help. Oh, the irony :) Cheers Jean-Michel. 2009/4/2 Gabriel - IP Guys gabr...@impactteachers.com: Dear All, Thanks for taking the time to read this. I have been presented with a massive task. I'm not an asterisk expert, but I do know my way around a linux server and infrastructure, and I know when things are not done correctly. A large number of minutes are routed every month, (1m+) and I wish to do this in the most efficient way possible. I've been presented with three linux servers, all in varying states of upkeep, and I've decided, instead of attempting to clean the systems I'm presented with, it is better for me to build a stable platform for asterisk to be migrated onto. This makes my question two fold. 1 What steps should I take, or consider, if I wish to migrate an existing asterisk installation, without it being offline for too long 2 What steps should I look out for, if I wish to move to a MySQL backed for the configuration files, so that I can remove the systems dependence on local configuration. My long term plan is to introduce MySQL to be the backend for the configuration and call log data and put this machine behind a load balancer, so that in due course, when I need to add more machines to handle the load, I will have no need to reconfigure asterisk, or build new configurations, and if I keep the base OS install uniform, I should in theory be able to deploy more asterisk boxes very fast behind a load balancer to increase the capacity of my VoIP Farm with minimal work. *VoIP farm is my term, please do not use it in any presentations to the powers that be inside your organisation - If you wish to do so please send £10(ten) via paypal to my email address which is clearly displayed in the email headers!* Also, in theory, it allows for testing of new configuration, without having to change the configuration on multiple machines at the same time. Which is always a good thing. Any help an advice, or questions are most welcome, as I wish to turn this mountain into a mole hill, a very stable, and expandable mole hill! Thank you for your time, Mr Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jean-Michel Hiver - Synapse co-founder CTO GSM +262 692 828 070 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPkall
Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk – who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. I get an ugly 404 when trying to sign up or log in... That is probably abandonware... :( ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk – who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). The sign up link doesn't work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
None of their pages apart from the front page seem to work though http://phone.ipkall.com/ipphone/login.asp Are you sure they still exist? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP Sent: Monday, April 06, 2009 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IPkall IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Call forwarding services for corporate users
Hello, For corporate users, how would you define Call Forwarding services ? 1. Would offer option A or B ? option A: no forwarding immediate busy no answer option B: no forwarding immediate busy no answer busy or no answer I've seen legacy PBX offering B and SIP phones offering A. Which is the trend ? 2. If an extension is unconditionally forwarded (immediate forwarding), and destination is either busy, not answering, ... it's possible to leave voicemail into original or forwarded extension, or leave no message at all. Which option would be more natural ? C: leave voicemail into original extension's voicemail D: leave voicemail into forwarded extension's voicemail E: leave no voicemail at all F: have it user configurable Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
On 6 Apr 2009, at 14:32, Dean Collins wrote: None of their pages apart from the front page seem to work though http://phone.ipkall.com/ipphone/login.asp http://phone.ipkall.com/login.asp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Hmm, this seem to be the biggest non cisco device i found as well, The breakout is a FXS, 50-pin Telco to rj11 converter ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
On Mon, 6 Apr 2009, ContactTel Business wrote: Hmm, this seem to be the biggest non cisco device i found as well, The breakout is a FXS, 50-pin Telco to rj11 converter ? Yes. MP-124 is a solid, stable device. Any vendor that sells you the MP-124 will have a breakout box (or patch panel) to sell you to get you your RJ-11 jacks. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Grandstream GXW4024 IP Analog Gateway Also seem to do it, not sure what is better between AC and GS.. I think AC is more complicated to program but better quality, while GS is half the price of the AC. But also comes with rj11 jacks.. the AC has a 50 pin Any opinions ? Basically need to wire 10 hotels ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
I registered few days back and got a DID. Maybe this is temporary ? On Mon, Apr 6, 2009 at 7:05 PM, Steve Howes st...@geekinter.net wrote: On 6 Apr 2009, at 14:32, Dean Collins wrote: None of their pages apart from the front page seem to work though http://phone.ipkall.com/ipphone/login.asp http://phone.ipkall.com/login.asp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
On Mon, 6 Apr 2009, ContactTel Business wrote: Grandstream GXW4024 IP Analog Gateway Also seem to do it, not sure what is better between AC and GS.. I think AC is more complicated to program but better quality, while GS is half the price of the AC. But also comes with rj11 jacks.. the AC has a 50 pin Any opinions ? Basically need to wire 10 hotels ;) Depends how often you want to visit those hotels to support their equipment. I've had nothing but problems with Grandstream. I've left Audiocodes boxes at installations and not touched them for four years. My Grandstream experience is several years old at this point, however. Perhaps they are more stable now. I was constantly rebooting the few 8 port boxes I tried to use back then, and voice quality was not good. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Actually thinking about it, that 50 pins is simply the 48 + 2 grounds i imagine.. or something of the likes.. Thanks Jeff, you have pointed me in the right direction. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Hmm, this seem to be the biggest non cisco device i found as well, The breakout is a FXS, 50-pin Telco to rj11 converter ? Yes. MP-124 is a solid, stable device. Any vendor that sells you the MP-124 will have a breakout box (or patch panel) to sell you to get you your RJ-11 jacks. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs On Mon, 6 Apr 2009, ContactTel Business wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would need to add a breakout box to get your RJ11 jacks). j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
Daniel Nowacki wrote: SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. I get an ugly 404 when trying to sign up or log in... That is probably abandonware... :( No. It's just poorly-checked web management. http://phone.ipkall.com/ Is the signup link. The /ipphone stuff appears to be an old document tree that no longer exists. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Douds it
I have a few questions. Asterisk is a windows program why each time I try to find out how communicate with my Panasonic TDA 100 or with TDE 100 always read use one card o use a box why I can't use simply my network card, in the other side of Panasonic exist two types of cards one in TDA 100 with 2 trunks and in the other side TDE have internal Two trunks too. Why if I want to use asterisk in home and I have a modem with voice I can't use it to access to a line. ? Apologize my English Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Douds it
jibanez1...@cimex.com.cu wrote: I have a few questions. Asterisk is a windows program Asterisk is not a windows program. why each time I try to find out how communicate with my Panasonic TDA 100 or with TDE 100 always read “use one card o use a box” why I can’t use simply my network card, in the other side of Panasonic exist two types of cards one in TDA 100 with 2 trunks and in the other side TDE have internal Two trunks too. If your panasonic allows SIP or h323 that might work. A quick google search shows the ethernet port is for configuration of the IVR only, so i guess that is a no. It does do TAPI,so if a tapi plugin for asterisk exists, this might work. Why if I want to use asterisk in home and I have a modem with voice I can’t use it to access to a line. ? Because you will need a driver for that card for asterisk. Apologize my English Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11’s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] async agi question
You have to understand that this mailing list is not free instant support. Even more, you are using an unsupported Asterisk feature for 1.4. I will check it when I have some spare time to try to reproduce and fix it. If you are too much in a hurry you can always contact me off-list for paid support for this feature. Moy On Mon, Apr 6, 2009 at 3:15 AM, Jose Arias cyr2...@gmail.com wrote: Hi, I was asked for the patch and I sent it. Does anybody have any news about this subject? I'm willing to try a fix for 1.4 but I'd need any guidelines to do it. Thanks in advanced Jose 2009/4/2 Moises Silva moises.si...@gmail.com Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something, do you still have the patch around? Moy On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote: Hi Henrik, I would like to do the same thing you are doing here. I want to implement an external queue functionality so I need to stop a play file launched previously with an async agi command on caller's channel, sending the call to agent's extension. I'm redirecting caller's channel with REDIRECT while playing is taking place but I'm always getting a hang up on caller's channel. I'm using: asterisk-1.4.18 asterisk-addons-1.4.7 async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4) Both caller and agent are using 501 and 500 extensions and the async agi loop is waiting on 800, for example. The caller is dialing 800 where a play file is commanded through and async agi stream file command by the application. The relevant part of extensions.conf follows: exten = _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN}); exten = _5.,n,Wait(1); exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto); exten = _5.,n,Hangup(); exten = _8.,1,Noop(every thing starting 8 ${EXTEN}); exten = _8.,n,AGI(agi:async); exten = _8.,n,Hangup(); And the redirect command the application is sending to is: Action: Redirect Channel: SIP/501-081f0730 Exten: 500 Context: sip_sercom Priority: 1 Therefore, Henrik, could you show me your related dial plan and the redirect command you are sending? I wasn't able to see what I'm getting wrong. thanks in advanced Jose M Arias -- This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Relay ringing sip message 180
Dears Asterisk is a median server between the caller and the terminations gateway The caller send the call to asterisk à asterisk will play music on hold untill the termination gateway send 200 OK and the RTP establish My problem that, Asterisk is not forwarding the 180 ringing from the termination gateway to the user How can I forward the sip message 180 to the caller or let the music on hold stop playing and the caller hears the Ring Back Tone which when 180 ringing from the termination gateway. What I am using now to stop the musinc on hold when the RTP established is exten = _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m) NB: I tried to edit chan_sip.c but I did not find the solution Please Advice Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Relay ringing sip message 180
On 6 Apr 2009, at 15:40, Khaled W. Chehab wrote: Dears Asterisk is a median server between the caller and the terminations gateway The caller send the call to asterisk à asterisk will play music on hold untill the termination gateway send 200 OK and the RTP establish My problem that, Asterisk is not forwarding the 180 ringing from the termination gateway to the user How can I forward the sip message 180 to the caller or let the music on hold stop playing and the caller hears the Ring Back Tone which when 180 ringing from the termination gateway. What I am using now to stop the musinc on hold when the RTP established is exten = _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m) NB: I tried to edit chan_sip.c but I did not find the solution Please Advice FFS stop posting the same crap over and over again, thats 4 times now ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Cisco Call Manager
On Sat, Apr 4, 2009 at 11:18 AM, Timothy Smith timotsm...@gmail.com wrote: We're migrating from Cisco to asterisk because cisco is expensive to maintain, besides we can achieve more with asterisk like customised IVRs etc. I don't know what expensive to maintain means. We spend more on our phone bill than what the gear costs by a significant margin. This being a large production environment, we can't just change over without testing thoroughly.. makes sense Now, i'd like to completely get rid of the cisco gateways by routing incoming calls through asterisk too (to the call manager, and finally the phones). I don't have an architecture diagram of your call flow, but you'll need to be picking off calls with some criteria, perhaps the number dialed or DNIS if these are PRIs and route accordingly. You should be able to make asterisk talk directly to your phones by putting some asterisk hardware onto the same address space as the phones. If these are Cisco phones most of them are multi-line, and you can go ahead and push a change that one of the lines will SIP-register with an asterisk system. At that point you don't need to route _through_ Cisco because you would be routing around Call Manager. If you want to keep Call Manager in the loop you need to have the traffic going to Cisco continue to look like it did before you put Asterisk in the loop, or you need to change the Call Manager to act according to the revised traffic it will be receiving from Asterisk. I recommend enabling call debugging on the Asterisk side and the Cisco side, and figure out which side(s) you want to change to keep both sides happy. After that, they'll be satisfied and i'll start registering the phones to asterisk until everything is asterisk. It has to be a smooth transition, just fyi, we're about 200 employees. I'll appreciate any advice towards achieving this. My advice is to test changes in a small lab setup before you cut them loose, assuming that no downtime is acceptable. No idea what your budget is, nor your type of call traffic. If this is real telco lines, you may need a test telco line for your experiments. (which you can also fake with asterisk and/or Cisco). If this is just SIP you can fake it with asterisk and cheaper used Cisco gear. Hopefully you already have a lab environment where you can test things out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Re: async agi question
cyr2...@gmail.com schrieb: This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com Use the appropriate header field for that information. It's called From (in contrast to Sender). Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Timer T309
Based on the Asterisk logs you posted the Asterisk doesn't have it implemented per: The implementation of timer T309 in the user side is optional Martin On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote: Martin escreveu: What is the specification for T309 ? I'm too lazy to look it up. The default behaviour when the alarm of layer 1 (electrical T1/E1) is detected is to assume all calls dropped on both sides and that's what Asterisk does. The timer is simply deactivated since all the calls are supposed to drop. I believe that agrees with Q921/Q931 specs. Martin On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote: Hi everione, I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco link in this scenario: Telco Phone -- Telco --- Asterisk Sip Phone When i make a call from Telco Phone to Sip Phone, the call complete, but when i disconnect the link and reconnect in few seconds, the Asterisk clear call: [Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 1: Red Alarm [Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on 'DAHDI/1-1' [Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 2: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 2: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 3: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 3: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 4: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 4: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 5: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 5: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 6: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 6: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 7: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 7: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 8: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 8: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 9: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 9: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 10: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 10: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 11: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 11: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 12: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 12: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 13: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 13: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 14: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 14: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 15: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Re: [asterisk-users] Please Advice SIP 183 progessl
Hi, The easiest is to turn off MOH on the Dial. Otherwise the patch is easy but not trivial. Once the B-leg receives the ringing message and passes it in Dial app then the code has to turn off the MOH and tell the A-leg to send the ringing message. At the same time the code that skips passing the ringing to A-leg has to be disabled. Martin On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab kche...@xplorium.com wrote: Dear Martin Can you inform me how to make the patch or from where I can get it otherwise if there is an application can generate it? Or if its relate to chan_sip.c ,please can you tell me which function to edit or lines to be added Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Sunday, April 05, 2009 5:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please Advice SIP 183 progessl Hi Khaled, app Dial clearly is coded to ignore the 180 Ringing being passed if you have 'm' option to Dial and you do. Try to take the 'm' out and see if 180 Ringing is passed to the A-leg. So if you want MOH and then when 180 Ringing comes to turn it off = you need a patch. Martin 2009/4/4 Khaled W. Chehab kche...@xplorium.com: 10x Martin , But B-Leg is sending 180 ringing Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IOS Interface
Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-topic: SIP DTMF most supported method
It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do anything else inband audio (only G711) Martin On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa cesc.sa...@gmail.com wrote: Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Registration and INVITE question
Have you looked at the syntax of register = keyword ? register = [transport://]user[:secret[:authuse...@host[:port][/extension] ; If no extension is given, the 's' extension is used. There you have it ... Contact: sip:s set the extension and you should be fine Martin On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote: I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to retrieve the calling party information
That's because you have to create a user account in sip.conf ... + Asterisk is sensitive about it. What should help is if you register the phone with that sip account first. Martin On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote: HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Registration and INVITE question
Thanks for the reply - Perhaps I was not clear. On the register= line, if I set /extension to be /12345, then this just replaces 's' with 12345, and ALL calls, regardless of their destination number will be routed on the INVITE line to 12...@x.x.x.x, and the actual destination is specified in the To: header. Not particularly useful, and I'd prefer not to have to go fumbling through the SIP headers to find what was really dialled :) Looking at the SIP RFC, the idea is that you specify a set of What I will accept details with each registration in the Contact: headers, which is intended to include _multiple_ possible destination addresses. The Registrar will then only ever send calls addressed to that list of destinations. Sadly, the RFC authors did not think to consider private point-to-point links where you can usefully say send me anything, you know best. Asterisk fills by defaulting to a single s...@x.x.x.x, where the 's' can be replaced by any single number. Most ITSPs work around this by assuming that they know best, and routing numbers even if they are missing from the registration. The odd exception does not do this. I suspect that the solution will be to register with a /extension of /pedanticitsp, and then have a dialplan which pulls and parses the SIP To: header. Other suggestions are still welcome. Regards, Steve 2009/4/6 Martin asteriskl...@callthem.info: Have you looked at the syntax of register = keyword ? register = [transport://]user[:secret[:authuse...@host[:port][/extension] ; If no extension is given, the 's' extension is used. There you have it ... Contact: sip:s set the extension and you should be fine Martin On Mon, Apr 6, 2009 at 7:45 AM, Steve Davies davies...@gmail.com wrote: I have an ITSP we are trying to work with that has an Unusual way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: sip:s...@x.x.x.x This then causes ALL calls from the ITSP inbound to us to be addressed: INVITE sip:s...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] In fact, whatever we send in the Contact: header is reflected in the INVITE for inbound calls, and the actual number dialled is always placed in the To: header. What 99.9% of our ITSPs would send is: INVITE sip:44123456...@x.x.x.x:5060;transport=udp SIP/2.0 To: sip:44123456...@x.x.x.x:5060;transport=udp [other headers omitted] As you can see, the correct destination number is placed into the INVITE header AND the To: header, and Asterisk routes it correctly based on the INVITE. My questions: - Is there a way of telling chan_sip to register with multiple Contact: headers in the registration request, so that all of the acceptable DDI numbers can be presented to the ITSP (This is what the RFC seems to suggest is the correct way to operate.) - Alternatively, has anyone encountered this previously, and perhaps created an s extension that then digs into the To: header, and routes according to that? Examples, workarounds and solutions are all welcome! Help? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Why would i want to do that ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Timer T309
Martin escreveu: Based on the Asterisk logs you posted the Asterisk doesn't have it implemented per: "The implementation of timer T309 in the user side is optional" Martin On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote: Martin escreveu: What is the specification for T309 ? I'm too lazy to look it up. The default behaviour when the alarm of layer 1 (electrical T1/E1) is detected is to assume all calls dropped on both sides and that's what Asterisk does. The timer is simply deactivated since all the calls are supposed to drop. I believe that agrees with Q921/Q931 specs. Martin On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote: Hi everione, I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco link in this scenario: Telco Phone -- Telco --- Asterisk Sip Phone When i make a call from Telco Phone to Sip Phone, the call complete, but when i disconnect the link and reconnect in few seconds, the Asterisk clear call: [Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 1: Red Alarm [Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on 'DAHDI/1-1' [Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 2: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 2: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 3: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 3: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 4: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 4: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 5: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 5: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 6: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 6: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 7: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 7: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 8: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 8: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 9: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 9: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 10: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 10: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 11: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 11: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 12: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 12: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 13: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 13: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 14: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 14: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 15: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on
Re: [asterisk-users] 25-50-100fxs
This may be your solution. Regards, Luis Morales On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business li...@contacttel.com wrote: Why would i want to do that ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11’s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Aint this based on asterisk ? I don't think i would use that, thanks anyway. And yes i know this is an asterisk list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs This may be your solution. Regards, Luis Morales On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business li...@contacttel.com wrote: Why would i want to do that ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Jorge Mendoza wrote: Brent Davidson wrote: Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an application, no related to Cisco OS. Jorge Mendoza Interesting. I've never seen anything refer to IOS other than in the context of the OS run by Cisco routers although with so many acronyms around I suppose it's just a given that some of them should have more than one meaning depending on the context. Anyway, as I've already gone way past my level of understanding on the subject I'll leave this thread to someone more qualified to weigh in. :-P ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Actually i might rephrase, i need hardware solution not pc based, no hard drives, no fans, no application you need to monitor, hence hardware, You can ignore the rest of this thread i have my info. Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel Business Sent: April-06-09 2:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 25-50-100fxs Aint this based on asterisk ? I don't think i would use that, thanks anyway. And yes i know this is an asterisk list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs This may be your solution. Regards, Luis Morales On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business li...@contacttel.com wrote: Why would i want to do that ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
David Ruggles schrieb: I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? Here's our source code (GPL) for provisioning of Grandstream phones: https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/htdocs/prov/grandstream/settings.php https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/htdocs/prov/grandstream/.htaccess Works for the BT 110, BT 200, GXP 1200, GXP 2000, GXP 2010, GXP 2020. Not sure about the GXV 3000. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOS Interface
Brent Davidson wrote: Jorge Mendoza wrote: Are there an IOS interface for Asterisk?, or an IOS to SIP converter? Some femtocells uses this protocol and I would to use them with Asterisk. Jorge Mendoza ___ You're comparing to apples to Orange. IOS is the Cisco operating system run by the Femtocells, not the protocol. I'm not familiar with Femtocells, but as far as I can tell (from reading wikipedia) they apparently can do SIP internally, but that is a more advance configuration and might require some additional software. Looks like they are more designed to what is called lub over IP which appears to be some sort of backhaul specification specific to Cellular / Wireless carrier technology. AFAIK, IOS interface for femtocell is a 3GPP2 specification. It is an application, no related to Cisco OS. Jorge Mendoza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP Voicemail - can't get messages. Arrgh!
Hi - I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now when I access an IMAP mailbox using VoicemailMain(), it always says there are no messages, even when there clearly are (unread) messages in the IMAP mailbox. I've also got the asterisk GUI running on this system, and its status page (retrieved via manager, I believe, or maybe voicemail show users) shows the correct message counts. I tried debugging manager messages to see how it was getting the message counts, but I didn't get any useful output. Does anyone know a better way (any way) to debug issues with IMAP Voicemail? I do see an error on the CLI: ERROR[20010]: app_voicemail.c:2026 mm_log: IMAP Error: Quota not available on this IMAP server Here's some background info: Asterisk: 1.6.0.8 IMAP Server: dovecot 1.0.7 c-client: UW imap2007e Config Files: voicemail.conf [general] format = wav49 serveremail = aster...@rosecompanies.com fromstring = ${VM_CALLERID} emailsubject = New voicemail. Length: ${VM_DUR} emailbody = ${VM_NAME}:\n\nYou have a new voicemail message. You currently have ${VM_MSGNUM} messages in your Inbox.\n\nFrom:\t\t${VM_CALLERID}\$ maxsecs = 600 minsecs = 4 skipms = 3000 maxsilence = 10 silencethreshold = 128 maxlogins = 20 userscontext = default imapserver = localhost imapfolder = INBOX authuser = asterisk authpassword = xxx maxgreet = 360 operator = yes maxmessage = 300 minmessage = 4 saycid = no sayduration = no envelope = no review = yes users.conf (a typical user): [02] username = 02 transfer = yes mailbox = 02 call-limit = 100 fullname = Test User cid_number = 02 context = DLPN_MainUsers hasvoicemail = yes vmsecret = xxx email = imapuser = allison hassip = yes hasiax = no host = dynamic nat = no hasdirectory = yes dtmfmode = rfc2833 threewaycalling = no callwaiting = no hasmanager = no hasagent = no canreinvite = no insecure = no pickupgroup = autoprov = yes label = 02 macaddress = 0004f200 linenumber = LINEKEYS = 1 secret = xxx disallow = all extensions.conf: exten = 000,1,VoicemailMain(s${CALLERID(num)}...@default) Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 25-50-100fxs
Mr. ContactTel, if you need hadware only take a look on: http://www.telephonydepot.com/Catalog/Digium-TDM2400P/Digium-TDM2400P-Blank-Board http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-S400M-Quad-FXS-Module http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-1U-Patch-Panel take a look over http://www.audiocodes.com/product-family/cpe-gateways Now you ever need an appliance o pc + asterisk solution. Regards, On Mon, Apr 6, 2009 at 1:56 PM, ContactTel Business li...@contacttel.com wrote: Actually i might rephrase, i need hardware solution not pc based, no hard drives, no fans, no application you need to monitor, hence hardware, You can ignore the rest of this thread i have my info. Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ContactTel Business Sent: April-06-09 2:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 25-50-100fxs Aint this based on asterisk ? I don't think i would use that, thanks anyway. And yes i know this is an asterisk list. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs This may be your solution. Regards, Luis Morales On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business li...@contacttel.com wrote: Why would i want to do that ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales Sent: April-06-09 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 25-50-100fxs Take a look on xorcom solutions http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded Regards, Luis Morales On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business li...@contacttel.com wrote: Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11’s jacks.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci
[asterisk-users] Sangoma and BT single lines
Hi, got a Sangoma A200 with a bunch of extension cards and having real problems getting it to deal with a normal single BT line Symptoms are that incoming calls are fine. Outgoing calls ring the far end, BUT asterisk never sees that the call is answered (ie no message in the logs files saying so), as a result the remove end can hear the PBX side talking, but there is no audio back from the remote side to us. When we hangup the log files show messages thave suggest it thinks the line is still ringing Comparing with another line which works fine (this is a BT multi-line system with what they call PBX signalling on it) I see that as soon as the remote end answers then asterisk gets a log message stating the same and audio is fine on this line Have now spent nearly 4 months trying to get the signalling sorted on this line. Most recently we requested dual signalling on the line - the end result is now that outbound calls work and asterisk reports that the phone answers, however, when you hangup the call then asterisk obviously gets a bunch of extra line reversals and things there is an immediate incoming call on the back of that outgoing call... Please - any suggestions on how to configure a Sangoma card for use with a normal BT single line? Thanks Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.0-rc4 Now Available
The Asterisk Development Team is pleased to announce the fourth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc4 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release candidate improves the performance of the ast_event cache functionality, fixes issues with rwlock corruption that would cause deadlock like symptoms, fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked, resolves an issue with dropped calls due to SIP re-INVITE 'glare', fixes the ability to retrieve voicemail messages from IMAP, and several other minor issues. This release also includes the change in the AST-2009-003 security advisory, which can be read at http://downloads.digium.com/pub/security/AST-2009-003.html For a full list of changes in this release candidate, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc4/ChangeLog Issues found in this release candidate can be reported at http://bugs.digium.com Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.9 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.9. Asterisk 1.6.0.9 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves a merge issue from trunk to 1.6.0.7 that caused memory to be freed that should not be. In trunk, pkt-data is an ast_str, but in 1.6.0, it is allocated in the same chunk of memory as the sip_pkt. Only the 1.6.0 branch of Asterisk is effected by this issue. For a summary of the changes in this release, please see the release summary: http://svn.digium.com/svn/asterisk/tags/1.6.0.9/asterisk-1.6.0.9-summary.txt For a full list of changes in this release, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.0.9/ChangeLog Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!
Noah Miller wrote: I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now when I access an IMAP mailbox using VoicemailMain(), it always says there are no messages, even when there clearly are (unread) messages in the IMAP mailbox. This appears to be the same issue as was resolved in bug 14685. If you use the latest version of Asterisk 1.6.0 branch then you shouldn't have that issue anymore. svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 asterisk-1.6.0-vanilla cd asterisk-1.6.0-vanilla ./configure make install Or wait for Asterisk 1.6.0.10 which will incorporate the changes since 1.6.0.8; the recent 1.6.0.9 release that went out only incorporates the changes to 1.6.0.7 plus the security release in 1.6.0.8, and a merge issue that crept into the 1.6.0 branch from trunk in 1.6.0.9. Thanks! -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hacked
Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Timer T309
Hi, You're right. I wasn't aware of this patch getting into the code. In the version you're running the code is already present. The only problem I see is that some other timer kicks in here and the T309 cannot be scheduled. q931.c has this ... /* For a call in Active state, activate T309 only if there is no timer already running. */ You'd have to probably dig deeper in it to find out more. But this is the latest explanation I see. That would explain why the call is disconnected/hanged up right when the alarm happens. One way to fix it for you would be to remove the already running timer so the T309 could be scheduled since anyways all other timers do not matter since without T309 the call is hanged up anyways. Martin On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann afo...@disc-os.org wrote: Martin escreveu: Based on the Asterisk logs you posted the Asterisk doesn't have it implemented per: The implementation of timer T309 in the user side is optional Martin On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote: Martin escreveu: What is the specification for T309 ? I'm too lazy to look it up. The default behaviour when the alarm of layer 1 (electrical T1/E1) is detected is to assume all calls dropped on both sides and that's what Asterisk does. The timer is simply deactivated since all the calls are supposed to drop. I believe that agrees with Q921/Q931 specs. Martin On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote: Hi everione, I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco link in this scenario: Telco Phone -- Telco --- Asterisk Sip Phone When i make a call from Telco Phone to Sip Phone, the call complete, but when i disconnect the link and reconnect in few seconds, the Asterisk clear call: [Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 1: Red Alarm [Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on 'DAHDI/1-1' [Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 2: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 2: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 3: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 3: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 4: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 4: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 5: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 5: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 6: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 6: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 7: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 7: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 8: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 8: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 9: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 9: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 10: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 10: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on channel 11: Red Alarm [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable to disable echo cancellation on channel 11: Invalid argument [Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected alarm on
Re: [asterisk-users] Hacked
http://www.websiteoutlook.com/www.songania.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann Sent: April-06-09 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hacked Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com _ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacked
Ok, I'll bite. What does websiteoutlook have to do with it? The IP mentioned is in the Netherlands: % Information related to '89.248.168.0 - 89.248.168.255' inetnum:89.248.168.0 - 89.248.168.255 netname:NL-ECATEL descr: AS29073, Ecatel LTD country:NL admin-c:EL25-RIPE tech-c: EL25-RIPE status: ASSIGNED PA mnt-by: ECATEL-MNT mnt-lower: ECATEL-MNT mnt-routes: ECATEL-MNT source: RIPE # Filtered role: Ecatel LTD address:Gyroscoopweg 2F address:1042AB Amsterdam address:Netherlands abuse-mailbox: ab...@ecatel.net admin-c:EL25-RIPE tech-c: EL25-RIPE nic-hdl:EL25-RIPE source: RIPE # Filtered % Information related to '89.248.168.0/24as29073' route: 89.248.168.0/24 descr: AS29073 route object origin: as29073 mnt-by: ECATEL-MNT source: RIPE # Filtered j On Mon, 6 Apr 2009, ContactTel Business wrote: http://www.websiteoutlook.com/www.songania.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann Sent: April-06-09 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hacked Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com _ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacked
ping www.songania.com PING www.songania.com (89.248.168.176) 56(84) bytes of data. 64 bytes from 89.248.168.176: icmp_seq=1 ttl=49 time=131 ms If you clicked on it you would of seen it shows info on the domain, that is hosted on it.. ill bite back ;) Then on bottom.. Owned By Al-Sharif Al-sharif ? rings a bell.. but who knows.. iptables --block all the worls minus what you want.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-06-09 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hacked Ok, I'll bite. What does websiteoutlook have to do with it? The IP mentioned is in the Netherlands: % Information related to '89.248.168.0 - 89.248.168.255' inetnum:89.248.168.0 - 89.248.168.255 netname:NL-ECATEL descr: AS29073, Ecatel LTD country:NL admin-c:EL25-RIPE tech-c: EL25-RIPE status: ASSIGNED PA mnt-by: ECATEL-MNT mnt-lower: ECATEL-MNT mnt-routes: ECATEL-MNT source: RIPE # Filtered role: Ecatel LTD address:Gyroscoopweg 2F address:1042AB Amsterdam address:Netherlands abuse-mailbox: ab...@ecatel.net admin-c:EL25-RIPE tech-c: EL25-RIPE nic-hdl:EL25-RIPE source: RIPE # Filtered % Information related to '89.248.168.0/24as29073' route: 89.248.168.0/24 descr: AS29073 route object origin: as29073 mnt-by: ECATEL-MNT source: RIPE # Filtered j On Mon, 6 Apr 2009, ContactTel Business wrote: http://www.websiteoutlook.com/www.songania.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann Sent: April-06-09 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hacked Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com _ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel Config
Is there different points in the zaptel configuration according to each country? Thanks. _ Drag n’ drop—Get easy photo sharing with Windows Live™ Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Config
I'd read this article (http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf) but as I see it, you only have 2 lines in zaptel.conf for country specification; the rest of the lifting is done in Zapata.conf. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, April 06, 2009 3:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zaptel Config Is there different points in the zaptel configuration according to each country? Thanks. _ What can you do with the new Windows Live? Find http://www.microsoft.com/windows/windowslive/default.aspx out ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!
I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now when I access an IMAP mailbox using VoicemailMain(), it always says there are no messages, even when there clearly are (unread) messages in the IMAP mailbox. This appears to be the same issue as was resolved in bug 14685. If you use the latest version of Asterisk 1.6.0 branch then you shouldn't have that issue anymore. Aha! Thanks, Leif! I'm not insane. OK, well, maybe I am. I didn't find that bug. I think I'm going to bite the bullet and go with 1.6.1.0-rc4. Some of those items in the 1.6.1.0rc4 changelog just look to good to be passed up (or too scary to ignore). Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Config
so they are only. loazone and defaultzone thanks. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 6 Apr 2009 16:01:25 -0500 Subject: Re: [asterisk-users] Zaptel Config I’d read this article (http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf) but as I see it, you only have 2 lines in zaptel.conf for country specification; the rest of the lifting is done in Zapata.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, April 06, 2009 3:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zaptel Config Is there different points in the zaptel configuration according to each country? Thanks. What can you do with the new Windows Live? Find out _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way AUDIO
I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
Can it be that any Port got blocked ? On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
How tcpdump on interface show?? 2009/4/6 David @ULC ucoms2...@gmail.com: Can it be that any Port got blocked ? On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Rubio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way AUDIO
Few Running figures !! On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote: I have a server with 2 Lan Cards. Now, when I am trying to make calls using Local Lan, its One way Audio which means customer cant hear me but if I use Static IP with Wan Connection, it works perfectly. I changed the network from loc1 to loc2 but its same. I tried changing Ethernet Card but no use. What could be the Issue ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacked
Can you give more information about this vulnerability ? Martin On Mon, Apr 6, 2009 at 2:55 PM, Jeremy Mann jm...@txhmg.com wrote: Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Australian NBN network announced
First posted at: http://deancollinsblog.blogspot.com/2009/04/australian-nbn-network.html You ripper, http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx This is exactly how fiber should be - a shared wholesale resource just like water to ensure retail channels can deliver value add in content and services; - Not 'outspending on infrastructure' disadvantaging the carrier or - Securing 'private monopolies' that disadvantage the customer. Lets hope this smashes the existing theory about how networks should be built. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to retrieve the calling party information
Thanks. Let me try it. On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote: That's because you have to create a user account in sip.conf ... + Asterisk is sensitive about it. What should help is if you register the phone with that sip account first. Martin On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote: HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to retrieve the calling party information
Yeah some devices use callerid as user which is xxx in x...@deviceip So if you see : chan_sip.c: Call from '1231231234' to extension '5544' rejected because extension not found. Then adding in the [user] stanza user=foobar fromuser=foobar insecure=very ( or port,invite if that still alive) Will make sure it can auth ok.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rilawich Ango Sent: April-06-09 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] fail to retrieve the calling party information Thanks. Let me try it. On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote: That's because you have to create a user account in sip.conf ... + Asterisk is sensitive about it. What should help is if you register the phone with that sip account first. Martin On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote: HI, Recently, I found that asterisk fail to get the correct context of the sip phone. Below is the configuration and the log message. In the log message, asterisk fail to identify the calling party. As a result, it use a default context instead of int. Anyone know why and how to fix it? (testing environment) asterisk 1.4.22 1.4.24 asterisk-addon-1.4.7 Setting name=123 context=int [Apr 6 17:53:12] NOTICE[4913] chan_sip.c: Call from '' to extension '5544' rejected because extension not found. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Australian NBN network announced
On Mon, 6 Apr 2009, Dean Collins wrote: http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx This is exactly how fiber should be - a shared wholesale resource just like water to ensure retail channels can deliver value add in content and services; - Not 'outspending on infrastructure' disadvantaging the carrier or - Securing 'private monopolies' that disadvantage the customer. Lets hope this smashes the existing theory about how networks should be built. They must have a different kind of government down under ;) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and patton
Hellow Can any body helps how can interfacing between asterisk and patton media getway. Thanks mahboob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and patton
Interface in what manner? mahboob zaman wrote: Hellow Can any body helps how can interfacing between asterisk and patton media getway. Thanks mahboob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users