Re: [asterisk-users] Internal CallerID problem
I am having the same issues when asterisk gets a call and then sends it to an Avaya system. Anyone have an idea as to what would be causing it ? On Nov 12, 2007 3:03 PM, Mark Bell [EMAIL PROTECTED] wrote: Hey Guys, I have something that just started happening. When my users call each other on their 5 digit extensions their CallerID is showing as [EMAIL PROTECTED] (X would be their Ext. and 10.25.2.50 is my server) Calls in an out to the outside world are fine. I have scoured my configuration and can't find what would be causing it. I have checked the sip.cfg in the polycom's and URI dialing is disabled. What am I missing? Trixbox 2.2.3 Asterisk 1.2.18 Polycom IP550's with 2.2 software Regards Mark ___ --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] ACD functionality , Skills for agents
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Tuesday, November 20, 2007 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD functionality , Skills for agents On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Beware of queue weights. They have caused major problems in the past for many people on this list. As I understand it, enabling weights requires * to grab a lock on a large number of data structures related to queue state, which can cause performance slowdowns and crashes. I haven't seen reports of this recently, so it might be better in the later 1.4 releases, but at one time it was a sure-fire recipe for pain. Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? It probably could be, but it would make reporting pretty difficult, as the key fields in the queue log are the call id and the queue name. While you could use the QueueLog() application to stick extra data about the call (e.g the skill chosen from the IVR) into the queue log, that would appear in one line only and require post-processing to glue it together with the rest of the data for that call. I'm pretty sure it wouldn't mesh nicely with the reporting package I use (QueueMetrics). KM: I'm actually using the same package (Queuemetrics 1.4.2)! What I do for this is maintain queue (skill) membership in a database, then add the channels to the appropriate queues when the agents log on via a web page. Is there a particular reason you want to just have one queue? KM: Well no if the ACD would work properly. As I mentioned there have been calls that were waiting in queue for 20 minutes because ACD was distributing calls from the rest of the queues with less waiting time. KM. ___ --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 1.4 spec file
On Tue, Nov 20, 2007 at 11:47:01PM -0800, Douglas Garstang wrote: Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Please provide feedback to bug http://bugs.digium.com/10950 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Softphone to be installed on the Mobile
Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. 2) If there is no wireless network, then it can receive calls via the GSM (doing a special settings on Asterisk to forward the call to the mobile number), so he can receive the call and do the PBX functionalies (transfer, pickup, forward)? I saw this in AVAYA, AL Catel, Cisco, ... Any help? Regards Bilal Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --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] chan_ss7 0.10.1
hi, i'm added another patch to chan_ss7 it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/ New in version 0.10.1 (community version) - support for more than 256 channels - zap style addressing http://download.seiros.ru/SeirosPBX/chan_ss7/ http://www.freevoice.cz/chan_ss7/chan_ss7-0.10.1.tar.gz md5sum a3ca3031f8f4ef96d505be3b297b47cc --- Marek Cervenka === ___ --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] Softphone to be installed on the Mobile
bilal ghayyad wrote: Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. 2) If there is no wireless network, then it can receive calls via the GSM (doing a special settings on Asterisk to forward the call to the mobile number), so he can receive the call and do the PBX functionalies (transfer, pickup, forward)? I saw this in AVAYA, AL Catel, Cisco, ... Any help? Regards Bilal Hi Bilal, someone mentioned to me yesterday something similar... They had a Bluetooth Dongle on their Asterisk box and when the Bluetooth enabled mobile came in range of *, calls would be routed to the mobile, once out of range, they would be routed to the Mobile phone number... For a softphone - I'd probably look for a Java based one. Don't most mobiles now run J2ME? HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ --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] Changing audio message to text message
Anthony Chapellier a écrit : Robert Lister a écrit : On Fri, Nov 16, 2007 at 02:28:45PM +0100, Anthony Chapellier wrote: Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? You might need to clarify what you are trying to do. When a call comes in for a particular queue, instead of playing an audio message in-band at the caller please hold the line you want to send some sort of text message somewhere... What sort of technology do you have in mind that you want to integrate? SMS? URL messages to other IP handsets? CTI integration with a web browser? pop-ups on user screens? Rob I'll clarify what I wish to do. I saw Asterisk was able to send an audio message You are in position 5, 30 min till someone answers you (something like that) to a person in queue. And since I'm working on a SIP softphone integrated in a web interface, I wanted to be able to show to softphone users the same message in text screened on the web interface while the audio message is still being sent. After some investigations, it seems I've got some ways to do so : - Add code to Asterisk to allow it to send the text messsage in SIP or in HTTP the same way it sends the audio message in RTP - Record queue logs in a mysql database and access it from the web interface - Execute the command line showing queue infos and find a way to get them (surely the less possible solution) But, maybe there are some other ways to do it ? I don't know... and I'm not sure to get what I want by chosing one of those solutions since I just want to get the position and the waiting time to be screened on the web interface. Does someone know how to remote access datas about average waiting time and caller position in a queue in real time ? are they stored in a database or a log file ? -- Anthony Chapellier - MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : [EMAIL PROTECTED] Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.com ___ --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] Softphone to be installed on the Mobile
Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --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] Softphone to be installed on the Mobile
On Wed, 21 Nov 2007, bilal ghayyad wrote: Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. You might want to look for phones that already have SIP clients built-in, rather than add-ons... Although the first company to produce an affordable IAX client for my Nokia E90 will get my money!) A lot of Nokia phones already have this in, and have had for some time. My E90 works OK with asterisk via Wi-Fi, but I've yet to be able to make it work via 3G... 2) If there is no wireless network, then it can receive calls via the GSM (doing a special settings on Asterisk to forward the call to the mobile number), so he can receive the call and do the PBX functionalies (transfer, pickup, forward)? I can ( do) transfer my incoming number to my mobile, then it goes over the traditional GSM network to the phone, but one into that network, you're at the mercy of that networks functions, so if the GSM network ( the phone!) lets you do transfers, etc. then you can... My E90 appears to let you make 2 SIP calls and transfer one to the other, but I've just tried it and it crashed... Gordon ___ --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] Softphone to be installed on the Mobile
try to use http://www.fring.com/download/ On Nov 21, 2007 3:28 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --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] blind transfer dumping calls
On Mon, 2007-11-19 at 09:14 -0500, Brian J. Murrell wrote: I am using asterisk 1.4.10 and seem to be having a problem with blind transfer. This could very well be a pebkac problem but I'm not sure. ... Called phone keys in '#': -- Started music on hold, class 'default', on channel 'Zap/1-1' -- SIP/1011002206-08245a80 Playing 'pbx-transfer' (language 'en') Called phone keys in '2005' -- Stopped music on hold on Zap/1-1 -- Transferring Zap/1-1 to '2005' (context internal-sip) priority 1 == Channel 'Zap/1-1' jumping out of macro 'dialhouse' Calling party hears click: -- Hungup 'Zap/1-1' And the call is dumped by that point. FWIW, an upgrade to 1.4.11 solved this problem. Cheers! b. ___ --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] Music on Hold Problem w/ Transfers
Lacy Brian, Could you please set verbosity to 10, then place your calls/holds/transfers and post the output? Both where it works and where it doesn't. Otherwise, helping you troubleshoot this will be difficult. Tony Plack On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. After posting, I dialed my cellphone, and music on hold works in all situations. It's something having to do with internal calls. I don't really care if that isn't working. I didn't think to try that first. ___ --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] ACD functionality , Skills for agents
Guys can someone answer how the ACD works when it needs to decide which call to take next from queues with equal weights? Does it take the call with the longest period of watiting or does it work randomly? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyriakos Sent: Wednesday, November 21, 2007 11:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ACD functionality , Skills for agents -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Tuesday, November 20, 2007 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD functionality , Skills for agents On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Beware of queue weights. They have caused major problems in the past for many people on this list. As I understand it, enabling weights requires * to grab a lock on a large number of data structures related to queue state, which can cause performance slowdowns and crashes. I haven't seen reports of this recently, so it might be better in the later 1.4 releases, but at one time it was a sure-fire recipe for pain. Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? It probably could be, but it would make reporting pretty difficult, as the key fields in the queue log are the call id and the queue name. While you could use the QueueLog() application to stick extra data about the call (e.g the skill chosen from the IVR) into the queue log, that would appear in one line only and require post-processing to glue it together with the rest of the data for that call. I'm pretty sure it wouldn't mesh nicely with the reporting package I use (QueueMetrics). KM: I'm actually using the same package (Queuemetrics 1.4.2)! What I do for this is maintain queue (skill) membership in a database, then add the channels to the appropriate queues when the agents log on via a web page. Is there a particular reason you want to just have one queue? KM: Well no if the ACD would work properly. As I mentioned there have been calls that were waiting in queue for 20 minutes because ACD was distributing calls from the rest of the queues with less waiting time. KM. ___ --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] ACD functionality , Skills for agents
I have often wondered the same thing. It seems to me to be random, or it works it out some way I am not familiar with. I have seen calls with wait time of 30 seconds get answered before calls with 30 minutes wait time from queues with equal weight. It would be great if someone who actually knows could answer or explain. Best regards, Örn Arnarson On Nov 21, 2007 2:15 PM, Kyriakos [EMAIL PROTECTED] wrote: Guys can someone answer how the ACD works when it needs to decide which call to take next from queues with equal weights? Does it take the call with the longest period of watiting or does it work randomly? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyriakos Sent: Wednesday, November 21, 2007 11:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ACD functionality , Skills for agents -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James FitzGibbon Sent: Tuesday, November 20, 2007 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD functionality , Skills for agents On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Beware of queue weights. They have caused major problems in the past for many people on this list. As I understand it, enabling weights requires * to grab a lock on a large number of data structures related to queue state, which can cause performance slowdowns and crashes. I haven't seen reports of this recently, so it might be better in the later 1.4 releases, but at one time it was a sure-fire recipe for pain. Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? It probably could be, but it would make reporting pretty difficult, as the key fields in the queue log are the call id and the queue name. While you could use the QueueLog() application to stick extra data about the call (e.g the skill chosen from the IVR) into the queue log, that would appear in one line only and require post-processing to glue it together with the rest of the data for that call. I'm pretty sure it wouldn't mesh nicely with the reporting package I use (QueueMetrics). KM: I'm actually using the same package (Queuemetrics 1.4.2)! What I do for this is maintain queue (skill) membership in a database, then add the channels to the appropriate queues when the agents log on via a web page. Is there a particular reason you want to just have one queue? KM: Well no if the ACD would work properly. As I mentioned there have been calls that were waiting in queue for 20 minutes because ACD was distributing calls from the rest of the queues with less waiting time. KM. ___ --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] Need help in selecting DTMF Mode
Hi here is my setup : 1. USER - PSTN - Asterisk A - IAX2 Trunk - Asterisk B - SER - Asterisk C (Accepting DTMF) All Asterisk box has dtmfmode = inband, when user pressed DTMF able to receive and working fine. 2. Asterisk C --- Dial Customer Customer input DTMF and its not taking any dtmf but If I change dtmfmode to auto Asterisk C will take DTMF from users but my first Scenario fails if I change dtmfmode = auto in Asterisk C. Need urgent help. Thanks Arun ___ --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] ACD functionality , Skills for agents
Kyriakos wrote: Guys can someone answer how the ACD works when it needs to decide which call to take next from queues with equal weights? Does it take the call with the longest period of watiting or does it work randomly? Whichever thread from the queue that does its processing first is the one that will get the next available member. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Bugtracker to use with Asterisk?
I have never tried doing this myself, but we use Bugzilla as a well-working bug tracking tool, and it has an import script called importxml.pl that can be used to import bugs using its own XML format. So you would likely need some kind of AGI to create the XML and then run importxml.pl for the bug to be automatically imported, and it sounds pretty simple to do Just my euro 0.02, l. On Tue, 20 Nov 2007 19:59:43 +0100, Vincent [EMAIL PROTECTED] wrote: Hello Now that I have my first IVR up and running, I'd like to have Asterisk create tickets in a bug tracker every time a call comes in. It's a nice way to know who's calling and why, before following up on the cause for the call. There are tons of bugtracking apps out there. Do you know of some that I should look at? Ideally, the interface shouldn't be much busier than JoS http://discuss.joelonsoftware.com/?joel . Thank you -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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 on Pcengines Alix board
Hi there we have astlinux running on alix board, it is awesome. Andrea Giuseppe Barichello ha scritto: Date: Mon, 19 Nov 2007 10:39:31 -0600 From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk on Pcengines Alix board To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Sun, 2007-11-18 at 22:14 +0100, Giuseppe Barichello wrote: I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. That's very interesting! I've been curious about trying this. Did you run across any challenges getting this setup? Two main issues: 1) Understanding how voyage linux configures read-only and rw mounts (I wanted to mount all /var tree as rw) 2) Getting MOH play MP3 sound files with Debian standard packages: I had to recompile Asterisk from source to fix it. Giuseppe ___ --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 __ Informazione NOD32 2674 (20071121) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini CTO - VoIP Gedam Europe S.r.l. - (Torino,Italy) Gedam Advanced Communication Ltd - (Dunedin,New Zealand) Strada da Bertolla all'abbadia di Stura, 151 - 10156 Torino - IT GSM. +39-329.1871756 - PSTN. +39-011.19824516 - FAX. +39-011.8338622 - http://www.gedameurope.com/ http://freevoip.gedameurope.com/ http://www.faropbx.com/ http://www.pbxsolution.net/ ___ --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] Problem installing Asterisk
I have installed Asterisk with FreeTDS many times before (this same Asterisk and same TDS version)... but today when I did the make it gave me this error: ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -DFREETDS_PRE_0_62 -c -o cdr_tds.o cdr_tds.c cdr_tds.c:82:2: warning: #warning You have older TDS, you should upgrade! cdr_tds.c: In function `tds_log': cdr_tds.c:208: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `mssql_connect': cdr_tds.c:326: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:326: error: (Each undeclared identifier is reported only once cdr_tds.c:326: error: for each function it appears in.) cdr_tds.c:326: error: `connection' undeclared (first use in this function) cdr_tds.c:349: error: too few arguments to function `tds_alloc_context' cdr_tds.c:375: warning: implicit declaration of function `tds_free_connect' cdr_tds.c:389: error: `result_type' undeclared (first use in this function) cdr_tds.c:389: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `tds_load_module': cdr_tds.c:429: warning: unused variable `result_type' make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/home/matth/asterisk126/asterisk-1.2.6/cdr' make: *** [subdirs] Error 1 We have freetds-0.64 installed, which is what we are running on all of our other servers. Any idea why only on this one would I get a TDS error on make? Linux 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux ___ --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] ACD functionality , Skills for agents
It would be nice to add an option of choosing to answer the call with the longest waiting time, or answer randomly, or round robin, etc... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, November 21, 2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACD functionality , Skills for agents Kyriakos wrote: Guys can someone answer how the ACD works when it needs to decide which call to take next from queues with equal weights? Does it take the call with the longest period of watiting or does it work randomly? Whichever thread from the queue that does its processing first is the one that will get the next available member. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Zaptel 1.4 spec file
On Wed, Nov 21, 2007 at 11:13:38AM +0200, Tzafrir Cohen wrote: On Tue, Nov 20, 2007 at 11:47:01PM -0800, Douglas Garstang wrote: Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Please provide feedback to bug http://bugs.digium.com/10950 I also forgot: while the userspace part of Zaptel is rather simple to package, packaging the kernel modules seems to require much more distro-specific voodoo. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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-dev] trunk working under windows!
I dont know any non-linux guys who use Cygwin. Drew Gibson wrote: but ... why? Zoa wrote: Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: As a result of the commit below, now trunk can be built and run under Windows/cygwin, including the building of modules. Haven't checked yet the functionality - some modules surely cause ill side effects or deadlocks on exit, so you need to play a bit with modules.conf . If you want to play with a very minimal version the following does something: ; -- modules.conf [modules] autoload=no load = res_monitor.so load = res_features.so load = chan_sip.so Unfortunately, loading other modules is a bit critical and depending on the order or the timing you get crashes etc. To build trunk under windows/cygwin you need at least the following pieces: bash binutils curl gcc libiconv minires (resolver library) libdb4.3(probably db4.2 too) and a bit of patience because the build takes around 15min or more. cheers luigi On Tue, Nov 20, 2007 at 04:12:11PM -, SVN commits to the Asterisk project wrote: Author: rizzo Date: Tue Nov 20 10:12:10 2007 New Revision: 89454 URL: http://svn.digium.com/view/asterisk?view=revrev=89454 Log: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. Modified: trunk/Makefile.moddir_rules trunk/apps/Makefile trunk/channels/Makefile trunk/pbx/Makefile trunk/res/Makefile Modified: trunk/Makefile.moddir_rules URL: http://svn.digium.com/view/asterisk/trunk/Makefile.moddir_rules?view=diffrev=89454r1=89453r2=89454 == --- trunk/Makefile.moddir_rules (original) +++ trunk/Makefile.moddir_rules Tue Nov 20 10:12:10 2007 @@ -66,9 +66,8 @@ ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) # linker options and extra libraries for cygwin SOLINK=-Wl,[EMAIL PROTECTED] -shared - LIBS+=-L../main -lasterisk -L../res + LIBS+=-L../main -lasterisk -L../res $([EMAIL PROTECTED]) # additional libraries in res/ - LIBS_RES:= -lres_monitor -lres_adsi -lres_features endif endif Modified: trunk/apps/Makefile URL: http://svn.digium.com/view/asterisk/trunk/apps/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/apps/Makefile (original) +++ trunk/apps/Makefile Tue Nov 20 10:12:10 2007 @@ -39,3 +39,9 @@ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules + +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_features.so -lres_ael_share.so -lres_monitor.so -lres_speech.so + LIBS+= -lres_smdi.so +endif + Modified: trunk/channels/Makefile URL: http://svn.digium.com/view/asterisk/trunk/channels/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/channels/Makefile (original) +++ trunk/channels/Makefile Tue Nov 20 10:12:10 2007 @@ -64,6 +64,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_monitor.so -lres_features.so +endif + clean:: rm -f gentone $(MAKE) -C misdn clean Modified: trunk/pbx/Makefile URL: http://svn.digium.com/view/asterisk/trunk/pbx/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/pbx/Makefile (original) +++ trunk/pbx/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,10 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + LIBS+= -lres_ael_share.so -lres_monitor.so +endif + clean:: rm -f ael/*.o Modified: trunk/res/Makefile URL: http://svn.digium.com/view/asterisk/trunk/res/Makefile?view=diffrev=89454r1=89453r2=89454 == --- trunk/res/Makefile (original) +++ trunk/res/Makefile Tue Nov 20 10:12:10 2007 @@ -25,6 +25,13 @@ include $(ASTTOPDIR)/Makefile.moddir_rules +ifneq ($(findstring $(OSARCH), mingw32 cygwin ),) + # order-only dependency: build res_monitor before res_features + res_features.so: | res_monitor.so + # res_features uses some functions from res_monitor + res_features.so_LIBS:= -lres_monitor.so +endif + ael/ael_lex.o: ael/ael_lex.c
Re: [asterisk-users] Problems with losing D-Channel on
I also intermittently get the Short write error followed by a cascade of Zap alarms, the funny part is I always get an alarm clear exactly 5 seconds after the first red alarm. The carrier always notes no drop. This happens on a variety of digium hardware, all connected to T1 PRI, some in NFAS. The fact that it happens on multiple cards, platforms, and configurations leads me to believe this may be a driver issue. Eric Delaporte wrote: Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route test with incoming and outgoing calls over all lines without any problem over the whole time. I also got a phone call with the providers service partner for the S2M part. He reset the line and putt he errorcounter to 0. After the test it was still on 0. When we plugged in the card again, there were again errors on the counter after ~ 5-10 minutes. After this, i put the asterisk, zaptel and libpri versions to newest versions, now it's working a bit better, but after 1 day fine work, it crashes again all calls. The system worked fine about 6 month now, but since 2 weeks I got the problems. Does anyone have any idea? The last points what is on my todo is to switch the pci slot. The cable which connects card to E1 interface is also switched. Kind regards, Eric I got a snippet from the console, where the problem is occuring. [Nov 16 15:57:09] ERROR[5499]: chan_zap.c:8178 zt_pri_error: Write to 39 failed: Unknown error 500 [Nov 16 15:57:09] ERROR[5499]: chan_zap.c:8178 zt_pri_error: Short write: 0/15 (Unknown error 500) [Nov 16 15:57:09] WARNING[5499]: chan_zap.c:3822 zt_handle_event: Detected alarm on channel 1: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 2: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 2 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 6: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 6 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 7: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 7 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 8: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 8 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 9: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 9 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 10: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 10 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 11: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 11 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 12: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 12 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 13: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 13 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 14: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 14 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 15: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 15 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 17: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 17 [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:6672 handle_init_event: Detected alarm on channel 18: Red Alarm [Nov 16 15:57:09] WARNING[2868]: chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel 18 [Nov 16 15:57:09] WARNING[2868]:
[asterisk-users] Help Dial extention
I have a Linksys sipura phone which does not dial ext 26 only, every other ext works. When I dial ext 26 it show to:0 instead. Does anybody know how to fix this? Thanks in advance. Jarga Jallow image001.jpg___ --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] [DB] Insert only one prefix for multiple numbers?
Hello Some of our customers bought a bunch of phone numbers whose prefix is the same, eg. 555-12xx - 555-1200, 555-1201, etc. There's a telco name for this, but I forgot what it's called (think it's DID in ISDN.) To avoid having to input all those numbers in the DB in the cidname group, is there a way to have Asterisk translate any such number to the same CID name? I'm thinking of something like 555-12??/Acme Inc. Thank you. ___ --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] Softphone to be installed on the Mobile
On Wed, 21 Nov 2007 01:29:24 -0800 (PST), bilal ghayyad [EMAIL PROTECTED] wrote: Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: I guess you're really looking for a (smart)phone that supports wifi in addition to GSM, and to which you can install an SIP client, provided it's not already there. The phone should be able to switch from GSM to wifi when it detects a wifi network strong enough. Although wifi/wimax is probably a good thing, I've heard they are still not good enough (drain batteries since they don't know how to switch to stand-by mode, etc.) ___ --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] Help Dial extention
Don't know if they are related, look for 26 on this page: http://www.freepbx.org/support/documentation/howtos/howto-resolve-freepbx-and-sipura-linksys-feature-code-conflicts -- On Nov 21, 2007 10:45 AM, Jarga Jallow wrote: I have a Linksys sipura phone which does not dial ext 26 only, every other ext works. When I dial ext 26 it show to:0 instead. Does anybody know how to fix this? Thanks in advance. ___ --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] ACD functionality , Skills for agents
Kyriakos wrote: It would be nice to add an option of choosing to answer the call with the longest waiting time, or answer randomly, or round robin, etc... Agreed, but, understand that each queue defined in app_queue is separate. The way the weights work is only by instructing a thread to go into another queue's data space (while holding a mutex lock to make sure multiple threads aren't walking on the same space) and make sure there aren't calls waiting where that queue has a higher weight than the one currently processing before it decides whether or not it can serve up calls to an available member. There is not one large, consolidated, pool of calls waiting for consideration when you are dealing with multiple queues in the current design of app_queue. As a result, true skills based routing with the existing app_queue is, difficult, at best. The queue application does a fairly good job for what most people need for it to do, but when you start getting into these more complex call/queue routing scenarios, you're defining a scope of requirements that the original app_queue just wasn't designed for. Features like queue weight were/are band aids to try to get you closer to the end run goal, but that band aid and others like it has come with its own costs as well (mutex deadlocks,etc) that many people here have complained about in the past. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Vicidial + Unicall mfcr2
Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well! Leonardo Silva Does Vicidial work together with Unicall/mfcr2 ? Best Regards -- Bruno de Assumpção Loureiro msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.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] Problem with AGI Script
Wow I can't believe I missed this, and I can't believe no one else saw it!
Look at the word FROM in both the script, and the way it is called. 'From'
and 'from'... that doesn't work.
On Nov 14, 2007 8:59 AM, Matt [EMAIL PROTECTED] wrote:
I have asterisk 1.2.18 running on a new system we just installed.
Although I've used AGIs many times in the past, I'm stumped on this one. It
may just be a simple issue that I need another eyeset to look at.
My AGI does the following:
#!/usr/bin/perl
#Load a few modules...
use Asterisk::AGI;
use DBI;
$AGI = new Asterisk::AGI;
#Grab input from Asterisk
my %input = $AGI-ReadParse();
#Some Debugging
$AGI-exec('SayDigits',$ARGV[0]);
exit;
All seems fine. If I run the script from the command line it works as
expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
EXEC SayDigits 333
However, when actually running in practice I get:
-- Executing AGI(Zap/23-1, GetEmailfromDID.agi|5706016716) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
-- AGI Script GetEmailfromDID.agi completed, returning 0
extensions.conf
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
exten = s,3,rxfax(${FAXFILE})
exten = s,104,Set([EMAIL PROTECTED])
exten = s,105,Goto(3)
Any thoughts on why asterisk doesn't seem to be passing anything to the
script and the script doesn't seem to be passing anything back? When I call
I do not hear the digits read to me, instead I just get thrown to the next
object after the digit reading.
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Re: [asterisk-users] Softphone to be installed on the Mobile
There's an application server that sits between asterisk and the gprs network that can switch calls real time between wifi, your office pabx extensions and the gsm network. I've forgotten the name of it but I remember it costs $US6,000 for 10 licenses. If you want me to find out more I can spend the time to look into it but only after you've said yes to the budget. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent Sent: Wednesday, November 21, 2007 10:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Softphone to be installed on the Mobile On Wed, 21 Nov 2007 01:29:24 -0800 (PST), bilal ghayyad [EMAIL PROTECTED] wrote: Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: I guess you're really looking for a (smart)phone that supports wifi in addition to GSM, and to which you can install an SIP client, provided it's not already there. The phone should be able to switch from GSM to wifi when it detects a wifi network strong enough. Although wifi/wimax is probably a good thing, I've heard they are still not good enough (drain batteries since they don't know how to switch to stand-by mode, etc.) ___ --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] Softphone to be installed on the Mobile
Divitas Networks http://divitas.com/ has an asterisk based solution that allows seamless roaming between the Wi-Fi and GSM network. An appliance connects to or is the PBX on the office LAN and a client runs on the smartphone. The appliance and client then coordinate which network to use based on signal strength and availability. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, November 21, 2007 1:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Softphone to be installed on the Mobile Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. 2) If there is no wireless network, then it can receive calls via the GSM (doing a special settings on Asterisk to forward the call to the mobile number), so he can receive the call and do the PBX functionalies (transfer, pickup, forward)? I saw this in AVAYA, AL Catel, Cisco, ... Any help? Regards Bilal __ __ Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --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] Problem installing Asterisk
On Wednesday 21 November 2007 09:09:13 Matt wrote: I have installed Asterisk with FreeTDS many times before (this same Asterisk and same TDS version)... but today when I did the make it gave me this error: ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr' We don't support version 1.2.6 anymore. That is a VERY old version. -- Tilghman ___ --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] Softphone to be installed on the Mobile
On Wed, Nov 21, 2007 at 11:35:42AM -0500, Dean Collins wrote: There's an application server that sits between asterisk and the gprs network that can switch calls real time between wifi, your office pabx extensions and the gsm network. I've forgotten the name of it but I remember it costs $US6,000 for 10 licenses. If you want me to find out more I can spend the time to look into it but only after you've said yes to the budget. There are several methods allowing you to do this, but most are operator dependent. UMA (unlicensed mobile access), which is an unfortunate name as in several countries (including all of EU) all spectrum is licensed (though some is license excempt - which isn't unlicensed). This uses a home basestation which connects back to the GSM operator over IP. There's basically a switch in the GSM core which can flip between the call over GSM or WiFi. In the UK BT call this Fusion (in conjunction with Vodafone). Companies like Truphone run a software shim on the phone. When you make a call it actually prefixes the outbound call with a Truphone prefix, so if it's via GSM the call actually goes through them. That way they get termination revenue from the mobile networks, which hopefully covers [1] the cost of the actual onward call. If in range of a WiFi network the phone establishes a connection to an end-point in Truphone's network and switches the call (and if the WiFi degrades, it switches back to GSM). Both the above assume that the calls terminate on a system controlled by the network (or Truphone), so that sessions can be controlled by them. It would be possible to do this with software in the phone and all calls terminating on your kit which then passes the call on to the PSTN, so if WiFi is available it can originate the call from there, but it needs to switch to GSM/etc if you move out of range - so you need to be in control of both end-points of the call. Steve Note [1] Truphone took T-Mobile to the High Court in the UK as T-Mobile refused to route their number block. They won and T-Mobile were forced to route Truphone's numbers. The court also ruled that as they were providing a VoIP service, they should only get sub-pence termination rates - which means they don't cover onward call rates (as they terminate to traditional telcos), so although Truphone won the battle, they pretty much lost the war. -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 as non-root/best practices
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only way I could get this working was to create a symbolic link - /usr/bin/asterisk to point to /home/asterisk .asterisk - using the link created in /usr/sbin/ would not work for 'asterisk -r' It seems that all commands in /usr/sbin/. were unexecutable by user 'asterisk' or 'admin' - I think that this is to do with the fact that the sbin directory is only designed for root executable files. What is your recommendation on having an admin user be able to edit configs without using the same username as the asterisk daemon - would you create a group 'asterisk' and have users 'admin' and 'asterisk' as part of that group - If the system was compiled to run as asterisk, then the owner for the config files are all stored in the /home/asterisk/ subdirectory and are owned by 'asterisk'. Can you offer any thoughts on that? Cheers :-) Robert Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. The daemon runs as the user asterisk. There is no reason why the admin should run as the user asterisk. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' - I added this:- asterisk ALL=/usr/sbin/asterisk -r NOPASSWD: ALL This is totally unrequired. You just need to set proper permissions for the socket /var/run/asterisk/asterisk.ctl . This is done in asterisk.conf - [files] ;astctlpermissions = 0660 ;astctlowner = root astctlgroup = asterisk ;astctl = asterisk.ctl http://svn.digium.com/svn/asterisk/branches/1.4/doc/asterisk-conf.txt asterisk ALL=/usr/sbin/safe_asterisk NOPASSWD: ALL Why would Asterisk need to run safe_asterisk? With an arbitrary parameter? You may want to permit some administrator to do that, but not the asterisk daemon. This probably opens the door to priviliges escalations. -- Tzafrir Cohen ___ --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] Aastra 480i CT - No Incoming Calls
I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny ___ --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] Queue Drops to Voicemail
Hello All, I am hoping someone out there can enlighten me on this issue. I am using asterisk 1.4.11. We have a call queue setup, and our agents log into the queue. As long as no one is on the phone the queue works properly. However, when there are agents on the phone, the queue will erratically drop calls to the queue. Any help will be extremely appreciated, and I will provide any conf files you may require. I have included excerpts of the config files I think you may need. Sincerely, Gregory Malsack Incoming line in extensions.conf exten = 8582294,1,answer() exten = 8582294,n,goto(csr|s|1) CSR context in extensions.conf [csr] include = default exten = s,1,answer() exten = s,n,Set(CDR(accountcode)=800) exten = s,n,Queue(802|n|||30) exten = s,n,background(csr) exten = s,n,queue(800) agents listed in users.conf (agent logs into extension 111 (normally)) [] callwaiting = no fullname = Agent 1 hasagent = yes hasdirectory = no hasiax = no hasmanager = no hassip = no hasvoicemail = no host = dynamic mailbox = secret = 1234 threewaycalling = no registeriax = no registersip = no canreinvite = no nat = no dtmfmode = rfc2833 disallow = all allow = all [111] callwaiting = no fullname = Conference Room hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 111 secret = 111 threewaycalling = no vmsecret = 111 registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = all allow = all All directives in agents.conf are remarked out. queues.conf [800] fullname = CSR Agent Queue strategy = rrmemory timeout = 8 wrapuptime = 20 autofill = yes autopause = no maxlen = joinempty = yes leavewhenempty = no reportholdtime = no musicclass = csr member = Agent/ member = Agent/1112 member = Agent/1113 member = Agent/1114 member = Agent/1110 member = Agent/1107 member = Agent/1138 member = Agent/1118 member = Agent/1149 ___ --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] Music on Hold Problem w/ Transfers
I'm having this problem. Here is my output with verbosity on 10: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2524-099012b0, SIP/2523|15) in new stack -- Called 2523 -- SIP/2523-09905220 is ringing -- SIP/2523-09905220 answered SIP/2524-099012b0 -- Packet2Packet bridging SIP/2524-099012b0 and SIP/2523-09905220 -- Started music on hold, class 'default', on SIP/2524-099012b0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2523-0990f110, SIP/2500|15) in new stack -- Called 2500 -- SIP/2500-09913080 is ringing -- Stopped music on hold on SIP/2524-099012b0 == Spawn extension (default, 2523, 1) exited non-zero on 'SIP/2523-0990f110ZOMBIE' -- Nobody picked up in 15000 ms [Nov 21 13:25:05] NOTICE[14600]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/2500-09913080' not posted -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2524-099012b0, u2500) in new stack -- SIP/2524-099012b0 Playing '/var/spool/asterisk/voicemail/default/2500/unavail' (language 'en') == Spawn extension (default, 2500, 2) exited non-zero on 'SIP/2524-099012b0' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 21, 2007 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold Problem w/ Transfers Lacy Brian, Could you please set verbosity to 10, then place your calls/holds/transfers and post the output? Both where it works and where it doesn't. Otherwise, helping you troubleshoot this will be difficult. Tony Plack On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. After posting, I dialed my cellphone, and music on hold works in all situations. It's something having to do with internal calls. I don't really care if that isn't working. I didn't think to try that first. ___ --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] Problem installing Asterisk
On Nov 21, 2007 11:45 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 21 November 2007 09:09:13 Matt wrote: I have installed Asterisk with FreeTDS many times before (this same Asterisk and same TDS version)... but today when I did the make it gave me this error: ake[1]: Entering directory `/home/matth/asterisk126/asterisk-1.2.6/cdr' We don't support version 1.2.6 anymore. That is a VERY old version. Sadly, it is one of the only stable versions. Is 1.4 even out of beta yet? I'm not aware that it is, yet it's being forced down people's throats. ___ --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] Aastra 480i CT - No Incoming Calls
Am Mittwoch, den 21.11.2007, 13:47 -0500 schrieb [EMAIL PROTECTED]: I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Hi Danny, I do not know what you are doing wrong, but you could check the following: - Is Do-not-disturb possibly activated on the phone? (just checking) - What does sip show peers say (in on-hook mode, is the phone correctly registered to asterisk?) - What does the CLI show when you call the phone from another extensions, with verbose somewhere around 10? - Might NAT have to do with it? Something special with the network socket you use for the phone? - You have a recent firmware on it, I guess. Better check that anyway. Best regards Anselm ___ --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] Caller ID Question
We just installed an Asterisk 1.4 system and have a Polycom 501 phone we are using to test it. We have a PRI installed as well and it works well. The problem When a call is incoming, the caller id says: 99 sip:[EMAIL PROTECTED] how do you get it to just say 99 and remove all of the rest? Thanks ___ --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] Music on Hold Problem w/ Transfers
On Wed, 2007-11-21 at 08:17 -0600, Tony Plack wrote: Lacy Brian, Could you please set verbosity to 10, then place your calls/holds/transfers and post the output? I had mine set to 100 in fact and did paste the portion of asterisk output where the transfer was happening. But as I said in a followup, I solved the problem by going to 1.4.11. Thanx though, for your input. b. ___ --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] Caller ID Question
I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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 as non-root/best practices
On Nov 21, 2007 12:37 PM, Robert McNaught [EMAIL PROTECTED] wrote: Thanks Tzafrir, I took the stuff out of visudo - it turns out the only way I could get this working was to create a symbolic link - /usr/bin/asterisk to point to /home/asterisk .asterisk - using the link created in /usr/sbin/ would not work for 'asterisk -r' It seems that all commands in /usr/sbin/. were unexecutable by user 'asterisk' or 'admin' - I think that this is to do with the fact that the sbin directory is only designed for root executable files. What is your recommendation on having an admin user be able to edit configs without using the same username as the asterisk daemon - would you create a group 'asterisk' and have users 'admin' and 'asterisk' as part of that group - If the system was compiled to run as asterisk, then the owner for the config files are all stored in the /home/asterisk/ subdirectory and are owned by 'asterisk'. Can you offer any thoughts on that? Cheers :-) Robert Robert, I don't see why a symlink is necessary... Try something like this: (as root) chown -R asterisk:asterisk /etc/asterisk chmod -R 770 /etc/asterisk usermod -G admin,asterisk admin Verify your /var/run/asterisk socket permissions as suggested by Tzafrir. The admin user should now be able to connect to the running asterisk socket and change the config files. Or, you could make /etc/asterisk mode 640, owned by admin:asterisk. Note because of the PATH for your admin user you will have to specify the full pathname to asterisk (usually /usr/sbin/asterisk). As far as permissions, you could add admin to the asterisk group and make sure your files are 660 (dirs 770). The various sbin paths are readable by all users, they just aren't in the PATH. Try this from your shell: as user: echo $PATH as root: echo $PATH Notice how the sbins are included while you are root but not while you are a user? That's because most of those binaries can't be used (at least not completely) by users other than root. Some distros that are setup for sudo (like Ubuntu) include sbin in most paths and expect you to use sudo. Please note the PATH variable in your shell is R/W and you can set it yourself. -- Kristian Kielhofner ___ --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 proxy failover
Hi, Is it possible to failover from Outbound SIP proxyA to Outbound SIP ProxyB on the event that Outbound Sip Proxy A became unavailable - using the qualify option for sip peers - it should be possible to monitor the ping/back time, which would give us a good indication of whether a host is up and running. I have had a look in the mailing list archives, but cant see this having been asked before? How would someone do this in Asterisk - would this have to be done with Dialplan programming, before placing the call, it would check the most recent qualify ping time and route based on that? As far as I am aware it is only possible to put host=xxx.xxx.com once in sip.conf Has anyone got this to work, to have a failover outbound proxy in asterisk, which automatically fails over? Thanks :-) Robert McNaught ___ --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] Aastra 480i CT - No Incoming Calls
I figured it out. Unlike the Linksys SPA942, the Web GUI interface for configuring the phone requires Proxy Server as well as the Registrar Server fields be populated with the IP address of the Asterisk server. [EMAIL PROTECTED] wrote: I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny ___ --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] [1.4 - Record] How to tell if user did leave a msg?
On Tue, 20 Nov 2007 23:27:34 -0500, Baji Panchumarti
[EMAIL PROTECTED] wrote:
use dialplan function STAT()
Thanks for the tip, but it doesn't seem to work:
==
exten = 888,1,Playback(/root/asterisk_sound_files/leave_msg)
exten = 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
exten = 888,n,Record(/tmp/${CALLTIME}.wav,3,30)
exten =
888,n,GotoIf($[${STAT(e,/tmp/${CALLTIME}.wav)}]?888,valid_msg)
exten = 888,n,Verbose(BAD!)
exten = 888,n,Hangup()
exten = 888,n(valid_msg),Verbose(HERE!)
==
Looks like Record() always creates the file, even if the user hung up
without leaving a message. Any other idea?
Thank you.
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Re: [asterisk-users] asterisk as non-root/best practices
On Wed, Nov 21, 2007 at 09:37:50AM -0800, Robert McNaught wrote: Thanks Tzafrir, I took the stuff out of visudo - it turns out the only way I could get this working was to create a symbolic link - /usr/bin/asterisk to point to /home/asterisk .asterisk - using the link created in /usr/sbin/ would not work for 'asterisk -r' It seems that all commands in /usr/sbin/. were unexecutable by user 'asterisk' or 'admin' - I think that this is to do with the fact that the sbin directory is only designed for root executable files. What is your recommendation on having an admin user be able to edit configs without using the same username as the asterisk daemon - would you create a group 'asterisk' and have users 'admin' and 'asterisk' as part of that group - If the system was compiled to run as asterisk, then the owner for the config files are all stored in the /home/asterisk/ subdirectory and are owned by 'asterisk'. Asterisk needs to be able to read those files. Not necessarily to write them You can also permit the admin user to write to the relevant config files using group ownership and permissions. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Caller ID Question
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
${CALLERID(num)})
${CALLERID(name)})
${DB(cidname/${CALLERIDNUM})})
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[asterisk-users] Error Unicall R2 Outgoing calls!!!
Hi, my name is Cristian, i am Argentina. I Have asterisk 1.4.11 with libs and patchs for unicall from http://www.moythreads.com/astunicall/. I work with mfcr2 and my configuration is: zaptel.conf span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone=us defaultzone=us unicall.conf [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4 protocolend=cpe group = 1 context= e1-incoming channel = 1-10 channel = 17-31 Incoming calls good, but can't outgoing calls, My hardware: TE110P. This is error: chan_unicall.c: Exception on 17, channel 1 Unicall/1 event protocol failure Unicall/1 protocol error. Cause 32769 Please help me Thank you!! Cristian. ___ --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 as non-root/best practices
On Wed, Nov 21, 2007 at 01:53:58PM -0500, Kristian Kielhofner wrote: (as root) chown -R asterisk:asterisk /etc/asterisk chmod -R 770 /etc/asterisk Nitpeeking: Now you made everything there executable. chmod -R o= /etc/asterisk chmod -R ug+rwX /etc/asterisk (Any shorter way?) In most cases there is also usually non real need for Asterisk to be able to write to its config files (except voicemail.conf and maybe a few others). usermod -G admin,asterisk admin -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 as non-root/best practices
Robert McNaught wrote: Thanks Tzafrir, I took the stuff out of visudo - it turns out the only way I could get this working was to create a symbolic link - /usr/bin/asterisk to point to /home/asterisk .asterisk - using the link created in /usr/sbin/ would not work for 'asterisk -r' It seems that all commands in /usr/sbin/. were unexecutable by user 'asterisk' or 'admin' - I think that this is to do with the fact that the sbin directory is only designed for root executable files. What is your recommendation on having an admin user be able to edit configs without using the same username as the asterisk daemon - would you create a group 'asterisk' and have users 'admin' and 'asterisk' as part of that group - If the system was compiled to run as asterisk, then the owner for the config files are all stored in the /home/asterisk/ subdirectory and are owned by 'asterisk'. Can you offer any thoughts on that? Cheers :-) Robert I'm not quite sure I understand where your troubles are... There are quite a few documented methods of building asterisk to run as a normal user, like on voip-info.org and my blog. If you follow the instructions, you should end up with an asterisk binary which runs as a non-root user and can access and write to the appropriate files where necessary and is pretty much invisible to the rest of the world, except root or admin if you prefer. Here's some links which describe the solution: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ --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] Caller ID Question
I have the same issue and I cant fix it :(
On Nov 21, 2007 9:56 PM, Vincent [EMAIL PROTECTED] wrote:
On Wed, 21 Nov 2007 12:54:22 -0600, Rob Schall [EMAIL PROTECTED]
wrote:
what cause's this? How do I get just 99?
Maybe there's a better way, ie. making the ISDN card or Polycom unit
handle the presentation, but you could have Asterisk rewrite the CID
name/number on the fly.
${CALLERID(num)})
${CALLERID(name)})
${DB(cidname/${CALLERIDNUM})})
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Re: [asterisk-users] Caller ID Question
Are you calling the other phones by URL or through asterisk? if your phone is registered to asterisk, and you ask to dial a number, it will connect through asterisk to another registered phone. If you ask to dial a url from the polycoms, i.e. sip:[EMAIL PROTECTED], then it will connect directly to the other SIP UA, skipping asterisk entirely. This is typically when you see the URL on the screen of the receiving phone. Am I clear? Sorry if I'm not. Mojo Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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] asterisk as non-root/best practices
Tzafrir Cohen wrote: On Wed, Nov 21, 2007 at 01:53:58PM -0500, Kristian Kielhofner wrote: (as root) chown -R asterisk:asterisk /etc/asterisk chmod -R 770 /etc/asterisk Nitpeeking: Now you made everything there executable. chmod -R o= /etc/asterisk chmod -R ug+rwX /etc/asterisk (Any shorter way?) This is how I did it: chown -R root:asterisk /etc/asterisk chmod 750 /etc/asterisk chmod 640 /etc/asterisk/* I also found I needed to make /etc/asterisk/voicemail.conf writeable by asterisk, so asterisk's configuration dorectory now looks like this: [ /etc/asterisk ]# ls -l total 44 -rw-r- 1 root asterisk 93 2007-10-14 11:54 asterisk.conf -rw-r--r-- 1 root asterisk 417 2007-10-16 21:34 codecs.conf -rw-r- 1 root asterisk 6645 2007-11-19 09:36 extensions.conf -rw-r- 1 root asterisk 958 2007-11-07 15:02 iax.conf -rw-r- 1 root asterisk 317 2007-10-04 16:01 logger.conf -rw-r--r-- 1 root asterisk 141 2007-10-25 13:17 meetme.conf -rw-r- 1 root asterisk 23 2007-10-04 16:01 modules.conf -rw-r- 1 root asterisk 1842 2007-11-05 21:11 sip.conf -rw-rw 1 root asterisk 1147 2007-11-19 09:55 voicemail.conf -rw-r- 1 root asterisk 737 2007-11-14 14:33 zapata.conf HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ --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] Caller ID Question
Disable URI dialing on your phones. CP Rob Schall wrote: I have an asterisk 1.4 setup with a PRI installed and working. We are using a Polycom 501 to test the setup.. Inbound calls work great as do phone to phone calls. However in all cases, the caller id is a bit odd. It shows: 99 sip:[EMAIL PROTECTED] what cause's this? How do I get just 99? Thanks ___ --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] [1.4 - Record] How to tell if user did leave a msg?
On Nov 21, 2007 2:51 PM, Vincent wrote: Thanks for the tip, but it doesn't seem to work: == [...] == Looks like Record() always creates the file, even if the user hung up without leaving a message. Any other idea? STAT() and record() are doing exactly what they are supposed to. Use the s flag to fetch the file size. You have to try a few hangups and figure out a minimum file size that qualifies as a recording in your setup. Based on the options you select for record you could have a file that is 25k in size but only has dead air, because that is how long it took to detect the hangup in your setup. Or you could have a 10k file with two words in it. Just depends. -baji. -- ___ --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] Error Unicall R2 Outgoing calls!!!
Try playing with the options. protocolvariant=ar,10,4,7 And please post debug output of unicall. unicall.conf loglevel=255 - Moy On Nov 21, 2007 2:04 PM, [EMAIL PROTECTED] wrote: Hi, my name is Cristian, i am Argentina. I Have asterisk 1.4.11 with libs and patchs for unicall from http://www.moythreads.com/astunicall/. I work with mfcr2 and my configuration is: zaptel.conf span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone=us defaultzone=us unicall.conf [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4 protocolend=cpe group = 1 context= e1-incoming channel = 1-10 channel = 17-31 Incoming calls good, but can't outgoing calls, My hardware: TE110P. This is error: chan_unicall.c: Exception on 17, channel 1 Unicall/1 event protocol failure Unicall/1 protocol error. Cause 32769 Please help me Thank you!! Cristian. ___ --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 -- Within C++, there is a much smaller and cleaner language struggling to get 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] Softphone to be installed on the Mobile
On 11/21/07, Arun Kumar [EMAIL PROTECTED] wrote: try to use http://www.fring.com/download/ I installed out of curiosity today, and guess what? You can do SIP over 3G (and probably wifi if you got it), plus the most unbelievable thing - you can talk and chat over Skype.. Even on Symbian S60.. (i wonder if they reverse-engineered the protocol, or bought it from somebody) Just two disatvantages - half-second lag even over 3G (maybe my provider is too slow), and that the battery of my N70 got drained over half-day.. guess i just have to buy second charger for work, but this really rocks :) Regards, Atis On Nov 21, 2007 3:28 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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] Problem dialing certain numbers with an E1 PRI
I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a similar problem when using Asterisk with R2, anytime you would dial a number belonging to another company you would get a busy tone. The way to solve it in R2 is to modify a timer T1 in mfcr2.c and increase the default value. I do not know if there is an equivalent when using ISDN. Here is the output from the CLI when making a call to one of those numbers: -- Executing [EMAIL PROTECTED]:1] Set(SIP/199-b7d023e0, TIMEOUT(absolute)=3600) in new stack -- Channel will hangup at 2007-11-21 23:15:05 UTC. -- Executing [EMAIL PROTECTED]:2] Dial(SIP/199-b7d023e0, Zap/g1/11070665||Ww) in new stack -- Making new call for cr 33190 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 422/0x1A6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 05 21 81 31 39 39] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '199' ] [70 09 a1 31 31 30 37 30 36 36 35] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '11070665' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 33190 on channel 1 enters state 1 (Call Initiated) -- Called g1/11070665 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3428 q931_receive: call 33190 on channel 1 enters state 3 (Outgoing call Proceeding) -- Zap/1-1 is proceeding passing it to SIP/199-b7d023e0 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: DISCONNECT (69) [08 02 84 9f] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3563 q931_receive: call 33190 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 31 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2716 q931_release: call 33190 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 422/0x1A6) (Originator) Message type: RELEASE (77) [08 02 81 9f] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Hungup 'Zap/1-1' [Nov 21 16:15:05] NOTICE[8991]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(SIP/199-b7d023e0, ) in new stack == Spawn extension (oficina-sup, 911070665, 3) exited non-zero on 'SIP/199-b7d023e0' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: RELEASE COMPLETE (90) Here is the config for that span: pbxarrgon*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part
[asterisk-users] spandsp as T.38 termination?
It seems that Spandsp has everything in it (when you include rxfax and txfax) to be a T.38 termination when used with Asterisk 1.4? And if so, what version of Spandsp? What version of IAXModem (so I don't have to also deal with T38Modem)? ___ --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] Music on Hold Problem w/ Transfers
Started music on hold, class 'default', on SIP/2524-099012b0 -- Please post your [default] section of musiconhold.conf Also need to know what version of Asterisk, version of kernel. Do you have ztdummy loaded (lsmod)? ___ --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] Aastra 480i CT - No Incoming Calls
There is a bug in the 480 firmware where if the callerid of the incoming call is malformed (or basically the Aastra doesn't like, for example have a # sign in the number), the phone won't ring. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 21, 2007 1:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Aastra 480i CT - No Incoming Calls I just bought an Aastra 480i CT for a client who needed cordless capabilities in their office. I'm trying to set up the base station and cordless handset in my office first. I'm able to connect the phone to my Asterisk box and make outgoing calls from either the base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny ___ --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] Problem dialing certain numbers with an E1 PRI
Carlos Chavez wrote: I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a similar problem when using Asterisk with R2, anytime you would dial a number belonging to another company you would get a busy tone. The way to solve it in R2 is to modify a timer T1 in mfcr2.c and increase the default value. It doesn't look like the problem you're having is caused by your local timers timing out. The other end is the one initiating the DISCONNECT. They send that before any of the local ISDN timers could time out. The cause is normal clearing, so it doesn't indicate really anything useful, just that it is hanging up the call out of nowhere. I do not know if there is an equivalent when using ISDN. Here is the output from the CLI when making a call to one of those numbers: -- Executing [EMAIL PROTECTED]:1] Set(SIP/199-b7d023e0, TIMEOUT(absolute)=3600) in new stack -- Channel will hangup at 2007-11-21 23:15:05 UTC. -- Executing [EMAIL PROTECTED]:2] Dial(SIP/199-b7d023e0, Zap/g1/11070665||Ww) in new stack -- Making new call for cr 33190 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 422/0x1A6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 05 21 81 31 39 39] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '199' ] [70 09 a1 31 31 30 37 30 36 36 35] Called Number (len=11) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '11070665' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 33190 on channel 1 enters state 1 (Call Initiated) -- Called g1/11070665 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3428 q931_receive: call 33190 on channel 1 enters state 3 (Outgoing call Proceeding) -- Zap/1-1 is proceeding passing it to SIP/199-b7d023e0 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: DISCONNECT (69) [08 02 84 9f] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the remote user (4) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3563 q931_receive: call 33190 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 31 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2716 q931_release: call 33190 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 422/0x1A6) (Originator) Message type: RELEASE (77) [08 02 81 9f] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Hungup 'Zap/1-1' [Nov 21 16:15:05] NOTICE[8991]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(SIP/199-b7d023e0, ) in new stack == Spawn extension (oficina-sup, 911070665, 3) exited non-zero on 'SIP/199-b7d023e0' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 422/0x1A6) (Terminator) Message type: RELEASE COMPLETE (90) Here is the config for that span: pbxarrgon*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active
Re: [asterisk-users] Music on Hold Problem w/ Transfers
Asterisk version 1.4.13 Also when I listened in on a transfer it sounds like the moh is trying to start but then immediately stop and tries to start again. Below is my musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh random=no -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 21, 2007 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold Problem w/ Transfers Started music on hold, class 'default', on SIP/2524-099012b0 -- Please post your [default] section of musiconhold.conf Also need to know what version of Asterisk, version of kernel. Do you have ztdummy loaded (lsmod)? ___ --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] spandsp as T.38 termination?
You need a T38 gateway of sorts, sort of like the app_t38gateway of CallWeaver. However digium refuses to include such a program with Asterisk. On Nov 21, 2007 6:13 PM, Robert Moskowitz [EMAIL PROTECTED] wrote: It seems that Spandsp has everything in it (when you include rxfax and txfax) to be a T.38 termination when used with Asterisk 1.4? And if so, what version of Spandsp? What version of IAXModem (so I don't have to also deal with T38Modem)? ___ --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] Vicidial + Unicall mfcr2
Hi Bruno, actually vicidial is working on top of asterisk, vicidial doesn't know what asterisk using in layer 2. SS7, ISDN stack, Unicall/mfcr2 is working with asterisk. vicidial uses asterisk application to deliver call center functionalities. Regards, Vidura. Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well! Leonardo Silva ___ --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] spandsp as T.38 termination?
On Wednesday 21 November 2007, [EMAIL PROTECTED] wrote: You need a T38 gateway of sorts, sort of like the app_t38gateway of CallWeaver. However digium refuses to include such a program with Asterisk. It's not a matter of refusal; it's a matter of licensing. We don't have any package that is free and clear. -- Tilghman ___ --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 support V5.2 protocal
Dear all anybody have idea about asterisk support V5.2 protocal ?? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org - Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how.___ --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] common/shared voicemail box
Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 5. Now, when someone calls 1000, and leaves a voicemail, I want to store the fact that this voicemail was meant for extension 1000. Similarly for 1001 and so on. Any ideas anyone? TiA - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
