Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Dan Austin wrote: John wrote: Thanks Steve for your suggestions. In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. This is exactly my current problem. NETCOM in Shanghai just told my local contact it is an E1 and that's it. I have no idea whether it is MFC/R2 or EuroISDN and so there is a lot of trial and error, not to mention about communicating with the telco. Is there anyway I could find out from zaptel what the line signal is? International installs are always fun. I have had some luck getting a local employee to relay my questions about provisioning, but all to often the response is 'We use the standard settings...'. At that point I resort to trial and error. I have setup a circuit in Shanghai, it is an E1, CRC4/HDB3 with the telco switch being/or compatible with ATT 5ESS. You should be able to get Netcom to tell you if the circuit is ISDN or not. Asking if it is a PRI will just confuse them, but they do understand the question 'ISDN or not ISDN' The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. That's great info, Steve. Just to comment - this is a great thread. I am expecting that the answer will either be quite interesting or quite odd. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
You don't need to install it. Just run kernel/xpp/utils/genzaptelconf directly from the source directory. Thanks Tzafrir. My local contact is away today and so I could not get him to plug the line to port 4. So, it is still in port 1. Here is the output after running genzaptelconf. # /usr/src/zaptel-1.4.11/kernel/xpp/utils/genzaptelconf # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF RED == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 B8ZS/ESF RED == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 B8ZS/ESF RED Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addressbook solution for Cisco 7961?
Andrew Latham wrote: Read my hacks on the Cisco phones in Oreilly's VoIP Hacks book Why the sales pitch for a 3 year old book? Can't you just give some information? Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addressbook solution for Cisco 7961?
Patrick wrote: Andrew Latham wrote: Read my hacks on the Cisco phones in Oreilly's VoIP Hacks book Why the sales pitch for a 3 year old book? Can't you just give some information? It's not a sales pitch - the book that he refers to is freely downloadable from the web. He /did/ give you some information - on exactly where to find the information you want. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whitepaper: How and to whom sell VoIP
Hello, Based on our own and our clients' experience we compiled short manual: How and to whom sell VoIP Hope it can be useful to some of you also. You can download it from our site: http://www.kolmisoft.com Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Queue: Code your own queue for AgentCallBackLogin
I am trying to build a simple queue with several agents using AgentCallBackLogin. From what I read on the Internet and tried briefly, it seems to suggest that I should be coding my own queue system for AgentCallBackLogin using AEL2 instead of using the AgentCallBackLogin command because it is buggy and will no longer be supported. Is this true? I don't seem to see too much literature on the Internet about using AEL2 or are people still waiting until we are forced to use AEL2? winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re-distributing E1
Before trying something impossible, and making a fool of my self, i rather ask the list... At my work i've got a single E1-test line, and all the project-leaders are constantly fighting over the use of the line. As it is mere the fact that they just need the E1 as a line, but not the amount of traffic, and the fact that they only needs it for several weeks or months, it is not worthwhile ordereing another line (for them) Can i redistribute the traffic of a single E1 transparantly over several other E1's? Was thinking about purchasing one or two quad E1 cards, andmapping incoming calls on cannel 1-5 to the first slave-E1, 6-10 to the second slave-E1 and so on... Just re-mapping B-channels. Most critical part is, that they should not see the difference between the original E1-line and the redistributed one's (besides the fact that their E1 will never be completely be filled) Is their any problem to expect with signaling (fco/fcx), timing and so on... Finally, i presume i need a couple of modem-pairs incase the line-length gets too long, not? hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] re-distributing E1
Yes you should be able to do that with an E-1 (it's called DACS). HOWEVER, you can't do DACS on a PRI, as you would need the D-Channel replicated and you can't do that. If you just want Asterisk to provide PRIs to your users, then that's a different story. Hans Witvliet wrote: Before trying something impossible, and making a fool of my self, i rather ask the list... At my work i've got a single E1-test line, and all the project-leaders are constantly fighting over the use of the line. As it is mere the fact that they just need the E1 as a line, but not the amount of traffic, and the fact that they only needs it for several weeks or months, it is not worthwhile ordereing another line (for them) Can i redistribute the traffic of a single E1 transparantly over several other E1's? Was thinking about purchasing one or two quad E1 cards, andmapping incoming calls on cannel 1-5 to the first slave-E1, 6-10 to the second slave-E1 and so on... Just re-mapping B-channels. Most critical part is, that they should not see the difference between the original E1-line and the redistributed one's (besides the fact that their E1 will never be completely be filled) Is their any problem to expect with signaling (fco/fcx), timing and so on... Finally, i presume i need a couple of modem-pairs incase the line-length gets too long, not? hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addressbook solution for Cisco 7961?
Rob Hillis wrote: Patrick wrote: Andrew Latham wrote: Read my hacks on the Cisco phones in Oreilly's VoIP Hacks book Why the sales pitch for a 3 year old book? Can't you just give some information? It's not a sales pitch - the book that he refers to is freely downloadable from the web. He /did/ give you some information - on exactly where to find the information you want. Do you perhaps have a link where to download it? All I can find is: http://oreilly.com/catalog/9780596101336/toc.html and it's not free but $29.95 so obviously I'm missing something. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HASH, HASHKEYS, ClearHash explanation
Tilghman Lesher wrote:
On Sunday 27 July 2008 13:28:00 Pavel Jezek wrote:
Hi, can somebody explain how to use this func/apps in asterisk?
I tried to find some examples on mailinglists or wiki, however without
success. thanks
The primary intended use is in conjunction with func_odbc, to allow you to
retrieve multiple values and reference them without having to assign each
field to an individual variable. Think 'SELECT *' where the fields included
might change over time, and you don't want the placement of a minor field
to completely break your dialplan.
thanks, can you give some example, if this functions can be used with
normal strings?
I found some example from bugreport 0008965, but I'm lost, what it's
actually doing ...
exten = s,n,Set(HASH(access_control)=${ACCESS_CONTROL_DATA(${USER_ID})})
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
emist wrote: My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. That was a bug that should have been resolved by 1.4.11 (he subsequently updated and it was resolved). Matthew Fredrickson Digium, Inc Hope it helps, Igor H. Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. If it is really R2, then maybe I need to buy an E100P card instead of TE412P. No, you should be fine with a TE412. Just make sure that your line is plugged in correctly and your span= line is correct for the line settings. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing calls authentication
Hello! I am looking for a configuration sample to authenticate outgoing calls. The idea is that each user have a password to dial any number. I was reading about Asterisk cmd Authenticate, Disa, etc. But I dont know how use this tools when I have running freepbx. Thanks for any idea. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing calls authentication
Gustavo A Gonzalez wrote:
Hello! I am looking for a configuration sample to authenticate
outgoing calls. The idea is that each user have a password to dial any
number. I was reading about Asterisk cmd Authenticate, Disa, etc. But
I don’t know how use this tools when I have running
exten = _9NXX,1,Set(RESTRICT=${DB(dialing/restricted)})
exten = _9NXX,n,GotoIf($[${RESTRICT} = YES]?3:7)
exten = _9NXX,n,Gosub(check_password,s,1)
exten = _9NXX,n,GotoIf($[${admin.afterhours} = Y]?5:3)
exten = _9NXX,n,Set(CDR(userfield)=${admin.password})
exten = _9NXX,n,Playback(auth-thankyou)
exten = _9NXX,n,Set(_ARG1=${CALLERID(num)})
exten = _9NXX,n,Gosub(set_callerid,s,1)
exten = _9NXX,n,Dial(ZAP/G1/${EXTEN:1})
exten = _9NXX,n,NoOP(${DIALSTATUS})
exten = _9NXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _9NXX,n,Hangup()
I have a cron job that locks the system with asterisk -rx 'database put
dialing restricted YES'
And the check_password looks like:
[check_password]
;***
;* Connect to SQL database to see if there is a match for **
;* the entry made by the end user **
;***
exten = s,1,Read(get-admin-password|enter-password|||3|)
exten = s,n,Gotoif($[${LEN(${get-admin-password})} 1]?10:3)
exten = s,n,MYSQL(Connect connid localhost username 'password'
Administration)
exten = s,n,GotoIf($[${MYSQL_STATUS} = -1]?mysql_failed,s,1)
exten = s,n,MYSQL(Query resultid ${connid} SELECT somestuff.aswell)
exten = s,n,MYSQL(Fetch fetchid ${resultid} somestuff.here)
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,Return()
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] ARA with MySQL or PostgreSQL
Hi everybody! I'm starting to do some test with Asterisk using Realtime Architecture. I would like to know your opinion about using MySQL or PostgreSQL in this schema. Which do you recomend? Are any benefits in any of them? Thanks in advance, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy Load Asterisk Array
Thanks for the reponse to both uf us. I'll be doing this soon, I hope. On Mon, Jul 21, 2008 at 9:25 PM, Jai Rangi [EMAIL PROTECTED] wrote: We also have the similar setup, 2 ser server with heartbeat doing the load balance and 4 asterisk servers handling the media. Of course the data is on MySQL Cluster. Jai Rangi www.bingotelecom.com On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote: I have used the OpenSer dispatcher module to load the calls (hash by caller id) to a group of asterisk boxes (In my case, 2 servers). The Asterisk boxes both use ARA and MySQL Master/Master replication. In a case like yours, I think you can use MySQL cluster, and you can still use Dispatcher to balance the load. On Mon, Jul 21, 2008 at 5:22 PM, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Creating Call permisions in Trixbox CE 2.6.1
Hi list, I've just downloaded and installed Trixbox CE (2.6.1) for first time and I want to replicate the functionality of my current Asterisk Setup (1.4.20.1) on it, but I cannot figure out how can I assign call permissions to a particular (group) of extensions, ie: who can make long distance call, international calls, mobile phone calls, etc. In regular (self compiled and configured without FreePBX) Asterisk I just create context for each kind of calls and include them in the context where the desired extensions resides, but I cannot find how to do this in Trixbox. I can make several Outbound Routes (one per call type) but, how can I restrict an extension from using any particular Outbound Route??? Thanks in advance, any help is really appreciated... -- Raul Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Code your own queue for AgentCallBackLogin
There's actually a document included with the source code which will take you through setting up an agent callback system. You can find it in 'doc/queues-with-callback-members.txt'. The 'AgentCallBackLogin' application has some issues, and since you can do the same thing with your dialplan, you're better off doing so. :M Lee, John (Sydney) wrote: I am trying to build a simple queue with several agents using AgentCallBackLogin. From what I read on the Internet and tried briefly, it seems to suggest that I should be coding my own queue system for AgentCallBackLogin using AEL2 instead of using the AgentCallBackLogin command because it is buggy and will no longer be supported. Is this true? I don't seem to see too much literature on the Internet about using AEL2 or are people still waiting until we are forced to use AEL2? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime still reads from .conf files
I've followed the instructions here ( http://www.voip-info.org/wiki-Asterisk+RealTime) and other places, however, Asterisk still reads information from the .conf files. How can I get Asterisk to read from the database and not from the .conf files? I realize the information above is sparse, but I do not know what other information is relevant. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARA with MySQL or PostgreSQL
On Wednesday 30 July 2008 11:19:29 Facundo Ameal wrote: Hi everybody! I'm starting to do some test with Asterisk using Realtime Architecture. I would like to know your opinion about using MySQL or PostgreSQL in this schema. Which do you recomend? Are any benefits in any of them? It depends on what you're doing with them. If you need fast read support, then MySQL is the way to go. However, if you're going to have a lot of write contention, or you're running heavy duty reports on the production server, then you'll want to be using Postgres. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIftime
Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) but i think that means July 30 22:00-2:00 and July 31 22:00-2:00 which looks like an invalid entry for the time. is it possible in asterisk? regards, Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIftime
Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) but i think that means July 30 22:00-2:00 and July 31 22:00-2:00 which looks like an invalid entry for the time. is it possible in asterisk? regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: GotoIftime
Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Nhadie Enviada em: quarta-feira, 30 de julho de 2008 16:47 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] GotoIftime Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) but i think that means July 30 22:00-2:00 and July 31 22:00-2:00 which looks like an invalid entry for the time. is it possible in asterisk? regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] It's telling me too much...
In case this is useful to others, a tip... I moved one of my Polycom 501's off it's subnet to another one (I've got an ether bridge glued to the back of the phone and a wireless card in the * box acting as AP). Now it is still served by the same Asterisk box, albeit through another ethernet port. It works just fine, except that on incoming calls, the full SIP address is displayed for the caller number. I was initially puzzled until I realized that the phone was simply qualifying the address of the caller because it was in a domain that is different than it's own - technically. But my users don't do technically, and come to think of it, I don't like it either. The fix: use iproute2 to mangle the packets: BEFORE: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro sh 192.168.20.0/24 dev eth0 proto kernel scope link src 192.168.20.3 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.99.1 71.245.116.0/24 dev eth2 proto kernel scope link src 71.245.116.10 169.254.0.0/16 dev eth2 scope link metric 1000 192.168.0.0/16 via 192.168.20.65 dev eth0 default via 71.245.116.1 dev eth2 metric 100 FIX: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro rep 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.20.3 AFTER: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro sh 192.168.20.0/24 dev eth0 proto kernel scope link src 192.168.20.3 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.20.3 71.245.116.0/24 dev eth2 proto kernel scope link src 71.245.116.10 169.254.0.0/16 dev eth2 scope link metric 1000 192.168.0.0/16 via 192.168.20.65 dev eth0 default via 71.245.116.1 dev eth2 metric 100 Now the phone thinks it's routing to the * box, but ignorance is bliss. Hope this helps someone. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIftime
Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIftime
At 01:36 PM 7/30/2008, you wrote: Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Doug Does that leave a 1 minute or 1 second hole? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk SIP Server/client connections
Hi Ken - The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. I just want to double-check the setup you have: you say the main server has the same config for each remote site connecting. Does that mean they're all connecting to the same SIP user/friend account? If so, that wouldn't work. You need to have a unique SIP account for each SIP device that's connecting. If that's not the case, and you have a unique sip account for each of your Polycom devices, can you show us the relevant part of your sip.conf from the main asterisk server? Also, do you get any particular messages on the console regarding this? Have you tried turning on SIP debugging? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Custom Filename for Incoming Agent Calls
Hi, to all, I have configured 3 Inbound/outbound agents queues, I record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I can
customize this filename recordings?
Thanks in advance.
Ricardo Melendez
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Re: [asterisk-users] GotoIftime
Ira wrote: At 01:36 PM 7/30/2008, you wrote: Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Doug Does that leave a 1 minute or 1 second hole? Should be neither. Since the time frame allowed is in minutes there is no time between 23:59 and 00:00 Since it has to be one or the other. Now, if I assume that the time is converted from time_t to 24 hour format using something such as localtime or gmtime the result of this should be using only tm_min and tm_hour which would also mean there is no hole. THESE ARE ALL ASSUMPTIONS, I have not checked the code. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIftime
On Wednesday 30 July 2008 14:46:55 Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) but i think that means July 30 22:00-2:00 and July 31 22:00-2:00 which looks like an invalid entry for the time. It's not invalid. That specifies July 30 midnight to 2am, July 30 10pm to midnight, July 31 midnight to 2am, and July 31 10pm to midnight. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIftime
cool. thanks for all your reply John Millican wrote: Ira wrote: At 01:36 PM 7/30/2008, you wrote: Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Doug Does that leave a 1 minute or 1 second hole? Should be neither. Since the time frame allowed is in minutes there is no time between 23:59 and 00:00 Since it has to be one or the other. Now, if I assume that the time is converted from time_t to 24 hour format using something such as localtime or gmtime the result of this should be using only tm_min and tm_hour which would also mean there is no hole. THESE ARE ALL ASSUMPTIONS, I have not checked the code. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIftime
On Wednesday 30 July 2008 16:19:52 John Millican wrote: Ira wrote: At 01:36 PM 7/30/2008, you wrote: Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Does that leave a 1 minute or 1 second hole? Should be neither. Since the time frame allowed is in minutes there is no time between 23:59 and 00:00 Since it has to be one or the other. Now, if I assume that the time is converted from time_t to 24 hour format using something such as localtime or gmtime the result of this should be using only tm_min and tm_hour which would also mean there is no hole. Actually, the timeframe is in intervals of 2 minutes, so he could have put 23:58, and it would have worked the same. The time is, in fact, set up into a bitfield, each bit representing a period of 2 minutes. If the current time maps into the bitfield to a location which is set, then the time matches. For the original question, note that the days of months, months, days of the week, and time are all checked independently, such that if any part fails its respective test, the whole thing fails. So even though *|*|30|* would specify an invalid date in February, it doesn't matter, as the algorithm would never find a 30th day in February, and so it would fail. Also note that GotoIfTime is unlike cron, whose algorithm for checking a match is: MINUTES and HOUR and ( ( DAY and MONTH ) or WEEKDAY ) the matching for GotoIfTime is: TIME and DAY and MONTH and WEEKDAY THESE ARE ALL ASSUMPTIONS, I have not checked the code. I helped write the code, and you are (mostly) correct. I just checked the code, just to be sure. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephan Weinberger Sent: 29 July 2008 15:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco vs Asterisk Am Freitag, 25. Juli 2008 18:54 schrieb Al Baker: This is simply NOT TRUE and shows a complete lack of understanding of modern software development. CISCO software is developed in a CMM environment. It has a formal test methodology and uses Automated Testing on EACH new release to ensure that 100% of the software that functioned in the Last Release, actually works in this release. This is simply NOT TRUE and shows a complete lack of understanding of software testing. Even if a company (or OS developer) implements automatic tests that cover each and every existing functionality this does NOT automatically ensure, that the existing functionalty works together well with new features. Nor does it ensure that existing code does work as intended under NEW circumstances. Testing (and I mean ANY form of testing, be it automatic or manual) can NEVER ensure the abscence of bugs! Additionally: Companies will never even atempt to find any bug. They will always only try to find as many bugs necessary to ensure that the cost of maintenance, bug-fixing, compensations and possibly loss of prestige does not exceed the cost of testing. Anything else would be financial loss. -- Stephan Weinberger Anyone who has watched the OSS vs Commercial Software debate over the last few years knows that OSS has turned the entire support model on its head and has questioned the rationale for paying huge sums for support so much so that even the commercial software companies have set up OSS-style communities to provide support for their products. I can tell you that you will most likely get an Asterisk bug fix before you get a CCM bug fix. As for bugs, perhaps you should check the list of known issues with CCM and compare with the list of bugs in Asterisk. All software has bugs, you just don't see a lot of people trying to get CCM to do things it wasn't designed for. - Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Filename for Incoming Agent Calls
On Wednesday 30 July 2008 16:09:11 Ricardo Melendez wrote:
Hi, to all, I have configured 3 Inbound/outbound agents queues, I record
Outgoing calls with custom filename like
outgoing-${callerid(num)}-${EXTEN}-${TIMESTAMP}.gsm
but I need to record Incoming calls and asterisk by default add 13 digits
number to inbound recordings like Agent-001-1298375678-890.gsm, how I can
customize this filename recordings?
Add one or two underscores in front of the variable name, to activate variable
inheritance, i.e.
Set(_MIXMONITOR_FILENAME=...) or Set(__MIXMONITOR_FILENAME=...)
--
Tilghman
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Re: [asterisk-users] Multiple Asterisk SIP Server/client connections
When I said same config I meant same with minor differences of account information :D [103] type=friend secret=1234 dial=SIP/103 callerid=Video103 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite [104] type=friend secret=1234 dial=SIP/104 callerid=Video104 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite I get no errors on the main server side, I get a lot of retransmitting messages on the remotes that don't get a connection. I'll post a portion of the log file later. The strange part to me is that it seems as if it's the first person in that wins. I can shutdown remote servers and bring them up individually and they work, but after that initial one it's as if the main server's ignoring port 5060 from other locations. I thought perhaps it was a firewall/router problem on the main server, so I swapped out a netgear router for a linksys wrt54g, same problem occurs on both routers. All 4 servers are listed as DMZ on their local firewalls. Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, July 30, 2008 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple Asterisk SIP Server/client connections Hi Ken - The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. I just want to double-check the setup you have: you say the main server has the same config for each remote site connecting. Does that mean they're all connecting to the same SIP user/friend account? If so, that wouldn't work. You need to have a unique SIP account for each SIP device that's connecting. If that's not the case, and you have a unique sip account for each of your Polycom devices, can you show us the relevant part of your sip.conf from the main asterisk server? Also, do you get any particular messages on the console regarding this? Have you tried turning on SIP debugging? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: i've installed several Asterisk systems in Shanghai Beijing. Thanks Edwin. The remote site is in Shanghai and NETCOM is the telco. Do you know if their E1 line is MFC/R2 or EuroISDN? i'm not sure if they provide MFC/R2. but we always ordered PRI from them. as far as switch type. seems like nobody in CNC can give us a definite answer, but we have success using EuroISDN swicth type. red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) As far as the cable goes, this is a bit complicated. The way it works is the telco delivers a fibre optic cable to the floor and the fibre terminates on a fibre optic multiplexer. Then the multiplexer is connected to a Fast Ethernet to E1 converter which has a RJ45 port. We then connect this RJ45 port to the TE412P port. Anyway what you said is still a good point - I will try replacing the straight through cable with a crossover and give it a go. if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. How should we complain? Are there any technical details we need to show them? It is a different country though. if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. p.s. note that T1/E1 crossover cable pin out is not the same as ethernet crossover cable. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HASH, HASHKEYS, ClearHash explanation
On Wednesday 30 July 2008 09:19:53 Pavel Jezek wrote:
Tilghman Lesher wrote:
On Sunday 27 July 2008 13:28:00 Pavel Jezek wrote:
Hi, can somebody explain how to use this func/apps in asterisk?
I tried to find some examples on mailinglists or wiki, however without
success. thanks
The primary intended use is in conjunction with func_odbc, to allow you
to retrieve multiple values and reference them without having to assign
each field to an individual variable. Think 'SELECT *' where the fields
included might change over time, and you don't want the placement of a
minor field to completely break your dialplan.
thanks, can you give some example, if this functions can be used with
normal strings?
I found some example from bugreport 0008965, but I'm lost, what it's
actually doing ...
exten = s,n,Set(HASH(access_control)=${ACCESS_CONTROL_DATA(${USER_ID})})
[CONTROL_DATA]
dsn=somedb
prefix=ACCESS
readsql=SELECT * FROM access WHERE user_id='${SQL_ESC(${ARG1})}'
--
Tilghman
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Re: [asterisk-users] Newbie Queue: Code your own queue for AgentCallBackLogin
Lee, John (Sydney) wrote: I am trying to build a simple queue with several agents using AgentCallBackLogin. From what I read on the Internet and tried briefly, it seems to suggest that I should be coding my own queue system for AgentCallBackLogin using AEL2 instead of using the AgentCallBackLogin command because it is buggy and will no longer be supported. Is this true? I don't seem to see too much literature on the Internet about using AEL2 or are people still waiting until we are forced to use AEL2? I have used addqueuemember and that works quite well with current versions of Asterisk and the old dialplan (ie: not ael2) I have the example code somewhere if you would like a copy. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
