Re: [asterisk-users] AstPligg
I'd frankly hoped that people here would be - on average - a bit smarter than hey! it's got a catchy name! and ajax buttons too!. l. In data Tue, 26 Jun 2007 03:00:48 +0200, Mark Phillips [EMAIL PROTECTED] ha scritto: Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --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 -- 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] Transfer Call to Cell Phone
by the way selling does not depend on the amount of lines you have and we are very productive trust me True, very true. There are lots of very productive sales people that don't need a phone at all. From the paper boy to car dealers, lots of sales don't require many phone lines. Of course at the same time, a typical call center wouldn't be very productive with only two lines. I have seen a million dollar corp work off four lines so your statement is quite vague... We have a few agents that have million dollar months and even a couple that have had million dollar weeks! But that isn't the point, is it Otis? The problem is you got your feelings hurt because instead of reading my reply, you assumed that I was putting your company down. My first paragraph was kind of a open thought process so that you and others might comprehend the basis of my reply. What I was trying to wrap my head around, was just how productive a system with only two lines could be if a single call came in and was then routed back out the other line to the outside sales guy. Now if you were using a digital line, perhaps we could consider using the signaling to redirect the call from the originating source directly to the salesman's phone and thus free up the lines for the next call. But no, you said pots lines as in Plain Old Telephone Service (POTS) which means we don't have the option of using some fancy, out of band signaling to redirect the call. So my thinking was, as I said before, Surely you have more than two lines. In my twenty-two years of telephony experience, dealing with everything from single line phones to key systems to PBX systems to Nortel DMS-500 switches, I only remember one sales office that only had two lines and that office was literally an 8 foot by 8 foot closet with two phones and all calls were outgoing. Yes, my answer was a little vague. So was the information you provided. Now had you bothered to read the 2nd and 3rd paragraph, you might have noticed that I provided a few methods that you could consider. My intention for doing this was simple. Maybe one of the ways mentioned would spark a response from you that would help to clarify the right way. Now suppose for a moment that you had actually read the reply. Let's also pretend that in reading it you realized that, yes, you have two pots lines, but what you had meant to say was that you had two unused pots lines along with some other form of incoming trunks. Then maybe you would have responded with an email to clarify that, to which I could have suggested that maybe you could look into a two port cell phone gateway to keep the incoming lines free and still keep connected to your sales guy. Can you see how we could have used that information to consider the right option? Considering that this list is for non commercial discussion, our only form of payment here is in the repayment of our debt to others that have gone before us and helped us out. Next time please appreciate the fact that someone else took time out of their busy day to consider and to reply your request for information. Now if you would like to provide a little more detail with your request, I'm fairly sure that someone here will likely respond to it. John ___ --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] Threading troubles 1.4.5 IAX2- SIP (FreeBSD specific??)
On 6/26/07, Jared Smith [EMAIL PROTECTED] wrote: I'm making a wild guess here, but I'd say that if you're using trunking, then you're probably getting close to exceeding the MTU size or possibly the MAX_TRUNKDATA size as defined in chan_iax2.c. If it's happening without IAX2 trunking turned on, then I have no idea what's happening... you'd have to look at the IAX2 and SIP packets when the problem is happening, and try to figure out what's causing the issue. It's not the trunking, as it disappears (Goes stable) when the threading is turned off... -- Hendrik Visage ___ --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] Best wifi IP phone for asterisk
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: HAve a look at the Linksys WIP 300 (or something) Can be charged from the USB port -- Hendrik Visage ___ --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] Rining 180 and 183
Replace with below. Actually Asterisk should only generate ringback when the SIP phone is ringing. On 6/25/07, satish patel [EMAIL PROTECTED] wrote: exten = 222,1,Dial(SIP/222,r) exten = 333,1,Dial(SIP/333,r) exten = 555,1,Dial(SIP/555,r) exten = 100,1,Dial(SIP/100,r) exten = 112,1,Dial(SIP/112,r) exten = 115,1,Dial(SIP/115,r) ___ --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] Inexpensive Layer 3 Switch?
Any recommendations on an economical layer 3 switch? I've been quite happy with the Netgear FS728TP ProSafe switches. These are 24 port 10/100 switches with 4 GigE ports and PoE. They also support 1000-SX GBIC through an optional module. The total PoE budget for all 24 port is 195 watts. We run 38 GXP2000 phones on a pair of them. We have one with 19 phones and a DLink DWL-3200AP wireless access point drawing power and the current load is under 75 watts average and has peaked at 97 watts. The other one has 19 phones, two Ethernet cameras and draws even less. All for less than $375 shipped. They even have a $25 rebate on them until the end of the month. Plus they even have a Lifetime Warranty. One of the cool features I discovered after installation was the built in Time Domain Reflectometer. The TDR is great for testing out the cables right after installation. We were able to use it to locate the screw that the drywallers intentionally ran through one of our cables. It was so obvious that the contractor paid to replace the cable. The only negative comment I have about them is the drone of the fans. Our wall mount rack is in the break room and the switches are easily the loudest item in the rack. They are not as loud as the Dell server we originally bought so if your using a 1U server from Compaq, Dell, HP, etc... you'll be fine. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.2.18 and HPEC
Any idea why I can't build HPEC for zaptel 1.2.18? It builds fine with 1.4.3... PaulH ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Legacy PBX
Hi all,
I have a isue with a Siemens Hicom conected to my asterisk, here is the
scheme:
Telco Asterisk --- Legacy PBX --- Legacy phones
The asterisk box has a TE210 (one PRI conected to Telco another PRI
conected to Siemens)
Everything works ok, but when I make an international call from legacy
phones to the telco, for example: 0034934452740, the Siemens only sends
to Asterisk the three first numbers 003.
Here is my config in extensions.conf:
[incoming-siemens]
exten = _X.,1,NoOP
exten = _X.,n,Dial(Zap/g2/${EXTEN})
exten = _X.,n,Hangup
[incoming-telco]
exten = _X.,1,NoOP
exten = _X.,n,Dial(Zap/g1/${EXTEN})
exten = _X.,n,Hangup
The other calls works great, incoming calls and outgoing calls. Any help
will be very apreciated, I'm a newbie doing this kind of asterisk
config, so any advice will be helpful.
Best regards,
Marc
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Re: [asterisk-users] CDR Records s as dst
It can be fixed with the patch from
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028093.html
Cheers, Troy
On 26/06/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = s,n,NoOp(CID_NAME : ${CID_NAME})
exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = s,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = s,n,NoOp(TRUNK : ${TRUNK})
exten = s,n,Set(CALLERID(name)=${CID_NAME})
exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
exten =
s,n,Set(PRESENTATION=${IF($[${CID_CLIR}=1]?prohib_not_screened:allowed_not_screened)})
exten = s,n,SetCallerPres(${PRESENTATION})
exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface
exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Playtones(busy)
exten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
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[asterisk-users] realtime_extensions
Hi now, I am using, realtime connection(mysql) for dialplan, but the following line must be added ,manualy to extensions.conf, before reloading.for each new context. [NEW_CONTEXT] switch = Realtime/@extensions is there any idea, to add this line to dbase too? thanks in advance Best MAni Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --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] Best wifi IP phone for asterisk
On Tue, 26 Jun 2007, Hendrik Visage wrote: On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: HAve a look at the Linksys WIP 300 (or something) Can be charged from the USB port I have a WIP330 and it doesn't work. Maybe it needs a firmware upgrade. Maybe it's a defective unit and all the others work fine. I haven't called support yet because I haven't had the chance. Audio in one direction cuts out completely for about 4 seconds every 10 seconds during the call. For 10 seconds it works fine... audio in both directions... then 4 seconds of silence in one direction... then 10 seconds of normal, etc etc, repeating forever. Completely unusable as-is. All my other Linksys IP Phones work great, though. I only have one WIP330. I don't have any WIP300's. My recommendation: Whatever you go with, buy ONE first for testing to make sure you're happy with it BEFORE you buy a boatload of them. -- Nick ___ --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-users Digest, Vol 35, Issue 92
Greetings! Due to high workload, I am currently checking and responding to e-mail twice daily at 12:00 PM EST and 9:00PM EST. If you require urgent assistance (please ensure it is urgent) that cannot wait until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867. Thank you for understanding this move to more efficiency and effectiveness. It helps me accomplish more to serve you better. Sincerely, Harold Riley ___ --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] VPN technology for snom 370
Hello, as from now on the snom 370 a special firmware exist to build secure VoIP-Infrastructures via OpenVPN http://openvpn.net-Technology. For further information go to http://snom.com/wiki/index.php/Networking/VPN Note: *That is a pre-release, probably the software is still unstable* cheers, Hirosh Dabui -- Hirosh Dabui computer engineering http://snom.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 1.4.1 app_addon_sql_mysql
It looks like you haven't install some mysql packages BEFORE make clean, make, make install. just install: libmysqlclient15-dev mysql-client mysql-server D - Original Message - From: Khaled Chehab To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Thursday, June 21, 2007 7:24 AM Subject: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql when I enter asterisk-addons-1.4.1 and make menuselect * Asterisk-addons Module Selection * Press 'h' for help. XXX 1. app_addon_sql_mysql [*] 2. app_saycountpl XXX 3. cdr_addon_mysql [ ] 4. chan_ooh323 [*] 5. format_mp3 XXX 6. res_config_mysql Cannot install app_addon_sql_mysql …. Any dependencies required ? Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- ___ --Bandwidth and Colocation provided by 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
[asterisk-users] Bridging two PSTN calls
Hi, I'm a novice trying out an experiment with Asterisk and was unsure of the hardware needs for it. I'm wondering if its possible to receive a call from an external number (PSTN) say A. Then make a call to another external number (PSTN) say B - and then bridge the two calls so that A is talking to B? What hardware will I need to be able to do this. Secondly, if I had x number of simultaneous calls (A talking to B) - how many PSTN lines would I need? I think 2x. I know this sounds ridiculously inefficient - but I think the experiment might open some avenues. Thanks, Doug Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --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] Query
Hi, I have put Digium TE120P card in PCI slot. So, lspci command gives the information in followimg format. 02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 11) Following modules are running when seen through lspci wcte11xp 22304 - ztdynamic 9804 - ztdummy 3468 - ip_conntrack_irc6640 - ip_conntrack_ftp7312 - ipt_state 1864 - iptable_mangle 2696 - ipt_REJECT 5160 - ipt_LOG 6280 - ipt_multiport 2376 - ip_conntrack 47524 - iptable_filter 2856 - ipt_limit 2280 - ip_tables 18168 - wcte12xp 44352 - zaptel180036 - but on running asterisk -vvvgc it stops by printing the following errrors '###' at line 41 of /etc/asterisk/zapata.conf Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown directive '###' at line 43 of /etc/asterisk/zapata.conf Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown keyword 'group' in trunkgroups Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register channel '1-31' Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module chan_zap.so failed! What is the problem actually can anybody tell me. Thanx and regards sanchal ___ --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] Test Message
Sorry to clutter up the mailiing list, but I've been unable to post to this list for the past 2 WEEKS! My ISP's blocking SMPT from other than his own servers. I think I've worked around it. - But if I see this message in the digest then I know I'm okay. Again. - Sorry for any inconvenience. Gary Guthary ___ --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] Digium TE120P setup problem [was: Re: Query]
Setting subject to a more descriptive one. On Tue, Jun 26, 2007 at 03:35:12PM +0530, [EMAIL PROTECTED] wrote: Hi, I have put Digium TE120P card in PCI slot. So, lspci command gives the information in followimg format. 02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 11) Following modules are running when seen through lspci lsmod, right? wcte11xp 22304 - ztdynamic 9804 - ztdummy 3468 - ip_conntrack_irc6640 - ip_conntrack_ftp7312 - ipt_state 1864 - iptable_mangle 2696 - ipt_REJECT 5160 - ipt_LOG 6280 - ipt_multiport 2376 - ip_conntrack 47524 - iptable_filter 2856 - ipt_limit 2280 - ip_tables 18168 - wcte12xp 44352 - This one is probably the one you actually need. zaptel180036 - but on running asterisk -vvvgc it stops by printing the following errrors '###' at line 41 of /etc/asterisk/zapata.conf Comments in asterisk config files begin with a ';', not a '#'. (/etc/zaptel.conf is not an asterisk config file and comments there do begin with '#'). Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown directive '###' at line 43 of /etc/asterisk/zapata.conf Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown keyword 'group' in trunkgroups Should this be in [channels]? Anyway, Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register channel '1-31' Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module chan_zap.so failed! What is the problem actually can anybody tell me. What is the output of: cat /proc/zaptel/* What do you have in /etc/zaptel.conf ? In /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[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] call fail from audiocode to sip trunk
Dear ALL
I have audiocode MP -124 with configure in asterisk Endpoint
configuration means every analog phone register in asterisk now thing is that i
have one more SIP trunk with mediant 2000
[auodiocode-mp-124]-[ * ]--[mediant 2000]-E1
When i call from audiocode MP -124 phone i got this error
-- Executing Dial(SIP/20-0889c4d8, SIP/mediant/1) in new stack
-- Called mediant/1
-- SIP/mediant-088a1a18 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/20-0889c4d8, ) in new stack
== Spawn extension (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8'
-- Executing Dial(SIP/24-0889c4d8, SIP/mediant/0) in new stack
-- Called mediant/0
my extension.conf file is
exten = 43,1,Answer
exten = 43,2,Dial(SIP/43)
exten = 43,3,Hangup
exten = 777,1,Answer()
exten = 777,2,Dial(SIP/777)
exten = 777,3,Hangup()
exten = 888,1,Answer()
exten = 888,2,Dial(SIP/888)
exten = 55,1,Dial(SIP/55)
exten = 66,1,Dial(SIP/66)
exten = _11.,1,Dial(SIP/mediant/${EXTEN:2})
exten = _11.,2,Congestion
what is the problem
-
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Re: [asterisk-users] Fax Throughput
Don Kelly wrote: I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. The longer transmit time is probably a result either (or both)... 1) retransmissions due to the audio being consistently corrupted, and ECM retransmissions to correct the corruption 2) training failure (probably due to corrupt audio) resulting in a slower transmission rate (e.g. 9600 bps vs 14400 bps) As to how to fix it... it's almost certainly audio degredation occurring in your Asterisk configuration or linkage... so debug your Asterisk setup. Lee. ___ --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] chan_features.so / asterisk 1.4.5
Carsten Bock schrieb: José Luis Ledesma schrieb: In my asterisk 1.4.5 chan_features.so has been installed properly... check in your asterisk-source if /channels/chan_features.so is present regards, Jack escribió: Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by asterisk 1.4.5? See the 1.4.5 Changelog: (http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.5) 2007-06-07 19:46 + [r68196] Olle Johansson [EMAIL PROTECTED] * channels/chan_features.c: Disable chan_features by default in menuselect Thanks for your answer and sorry for my late response. So what does this exactly mean to me? Can I keep chan_features.so from 1.4.4? What consequences does it have when chan_features.so is disabled und why has this been done? Is chan_features.so related to features.conf? Regards, Jens ___ --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] AstPligg
Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --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] Best wifi IP phone for asterisk (LINKSYS SUPPORT QUALITY)
I haven't been too impressed with the WIP330 - but my experience with Linksys tech support has been disastrous! I spent approximately 50 minutes on hold, I was transferred between 4 different people (all of whom had a poor grasp of the English language), none of them understood the features of the phone. All they could do was read the user manual I already had in front of me. None of them could explain the auto-provisioning feature mentioned on the Linksys website, or on the phone menus, etc. None of them even understood what SIP and RTP protocols were. They were just there to read the manual to me. Here's a warning for the group...watch our for Linksys! -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Seraphin Sent: Tuesday, June 26, 2007 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best wifi IP phone for asterisk On Tue, 26 Jun 2007, Hendrik Visage wrote: On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: HAve a look at the Linksys WIP 300 (or something) Can be charged from the USB port I have a WIP330 and it doesn't work. Maybe it needs a firmware upgrade. Maybe it's a defective unit and all the others work fine. I haven't called support yet because I haven't had the chance. Audio in one direction cuts out completely for about 4 seconds every 10 seconds during the call. For 10 seconds it works fine... audio in both directions... then 4 seconds of silence in one direction... then 10 seconds of normal, etc etc, repeating forever. Completely unusable as-is. All my other Linksys IP Phones work great, though. I only have one WIP330. I don't have any WIP300's. My recommendation: Whatever you go with, buy ONE first for testing to make sure you're happy with it BEFORE you buy a boatload of them. -- Nick ___ --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] No CID on Zaps - TDM400
I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs). With Trixbox out of the mix and a regular phone connected I get the CID fine yet Trixbox shows 'unknown': dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'ringall' Here is my Zapata.conf if it helps: # ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf # Thank You, Eric ___ --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] chan_features.so / asterisk 1.4.5
Jack wrote: Thanks for your answer and sorry for my late response. So what does this exactly mean to me? Can I keep chan_features.so from 1.4.4? What consequences does it have when chan_features.so is disabled und why has this been done? Is chan_features.so related to features.conf? chan_features.so doesn't provide anything useful, it's not used anywhere because it's not really finished. As for why it was changed to be disabled by default oej probably thought that since we aren't using it why do we have it enabled by default. It's fine to live in a world without chan_features.so :) -- Joshua Colp Software Developer Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] AstPligg
Whats wrong w/ voip-info.org? Jon Weisman | Sales Engineer International Bell Communications www.ibell.net - Original Message - From: Alex Robar To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, June 26, 2007 8:38 AM Subject: Re: [asterisk-users] AstPligg Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] -- ___ --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] Transfer Call to Cell Phone
John Faubion wrote: by the way selling does not depend on the amount of lines you have and we are very productive trust me True, very true. There are lots of very productive sales people that don't need a phone at all. From the paper boy to car dealers, lots of sales don't require many phone lines. Of course at the same time, a typical call center wouldn't be very productive with only two lines. I have seen a million dollar corp work off four lines so your statement is quite vague... We have a few agents that have million dollar months and even a couple that have had million dollar weeks! But that isn't the point, is it Otis? The problem is you got your feelings hurt because instead of reading my reply, you assumed that I was putting your company down. My first paragraph was kind of a open thought process so that you and others might comprehend the basis of my reply. What I was trying to wrap my head around, was just how productive a system with only two lines could be if a single call came in and was then routed back out the other line to the outside sales guy. Now if you were using a digital line, perhaps we could consider using the signaling to redirect the call from the originating source directly to the salesman's phone and thus free up the lines for the next call. But no, you said pots lines as in Plain Old Telephone Service (POTS) which means we don't have the option of using some fancy, out of band signaling to redirect the call. So my thinking was, as I said before, Surely you have more than two lines. In my twenty-two years of telephony experience, dealing with everything from single line phones to key systems to PBX systems to Nortel DMS-500 switches, I only remember one sales office that only had two lines and that office was literally an 8 foot by 8 foot closet with two phones and all calls were outgoing. You are right but finish reading thisTo be honest I did not get my feelings hurt, so assumptions are not needed I was simply stating one scenario where a local company here was very product to better your understanding of how some companies work off 2 to 10 lines and still produce. If you have read my first statement you would have understood that I did read and I did appreciate your reply. As the other methods has no interest to me at this time. I do agree maybe I should have sent a paragraph with details but I felt like knowing only two lines for the sales office was plenty. Now do not confuse us with a call center. Trust me by no means could we be as productive as others but then again we are not a giant but a small business and do not sign up thousands a day so we do not have a need for more lines yet. But there again you still made a judgment call about the two lines. If I tell this is the setup then there is nothing to question. Sometimes all tech guys have a problem with assuming sometimes there are more to a situation than what was presented. I am guilty as well. LOL!! Yes, my answer was a little vague. So was the information you provided. Now had you bothered to read the 2nd and 3rd paragraph, you might have noticed that I provided a few methods that you could consider. My intention for doing this was simple. Maybe one of the ways mentioned would spark a response from you that would help to clarify the right way. Now suppose for a moment that you had actually read the reply. Let's also pretend that in reading it you realized that, yes, you have two pots lines, but what you had meant to say was that you had two unused pots lines along with some other form of incoming trunks. Then maybe you would have responded with an email to clarify that, to which I could have suggested that maybe you could look into a two port cell phone gateway to keep the incoming lines free and still keep connected to your sales guy. Can you see how we could have used that information to consider the right option? Here again had you read my first statement you would have understood that I appreciated your reply. John thanks for the input. forget about my right way ok! I made a mistake on putting this in...this is what I was really looking for: explained later down.. Considering that this list is for non commercial discussion, our only form of payment here is in the repayment of our debt to others that have gone before us and helped us out. Next time please appreciate the fact that someone else took time out of their busy day to consider and to reply your request for information. Now if you would like to provide a little more detail with your request, I'm fairly sure that someone here will likely respond to it. There again making assumptions are not right because I did try your first option. But John do not get me wrong as I have been thinking about the second so before you can say please appreciate the response lets try to get the facts straight.
Re: [asterisk-users] AstPligg
I don't think anything is _wrong_ with VoIP-Info at all, I just think the sites serve different purposes. This is all just personal preference, but to me VoIP-Info does not work that well as a social news site, as all stories/headlines, good or bad, have equal weight. With the Pligg system, the stories that are better are usually voted up so they have higher exposure. VoIP-Info is a great site for sharing Asterisk recipes and HowTos, but I'm not a fan of it as a news site. AR On 6/26/07, Jon Weisman [EMAIL PROTECTED] wrote: Whats wrong w/ voip-info.org? Jon Weisman | Sales Engineer International Bell Communications www.ibell.net - Original Message - *From:* Alex Robar [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Tuesday, June 26, 2007 8:38 AM *Subject:* Re: [asterisk-users] AstPligg Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! -- Alex Robar [EMAIL PROTECTED] ___ --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] Multi port IAX Gateway
I am looking for a gateway that has several FXS ports and uses IAX. I have a need for 16 ports, but will accept 6 or 8 port gateways as well. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.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] Inexpensive Layer 3 Switch?
You do not need an L3 switch for this, just any managed switch which does vlans Unless there is something else? On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I’d be glad to hear them…so far I’m looking at the SFE2000 from Linksys. thanks No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 PM ___ --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] Fax Throughput
Can you post your zaptel.conf so we can verify your timing settings? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 25, 2007, at 11:10 PM, Don Kelly wrote: I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. Any suggestions? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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] rcf2833 DTMF broken in asterisk SIP channel?
I posted this bug yesterday: http://bugs.digium.com/view.php?id=10058 but really was hoping that one of you would be willing to try something simple for me and reply back with your results. Basically - I have run into a problem where Asterisk RFC2833 DTMF does not seem to be compatible with large SIP providers such as Level 3 and Global Crossing. Can someone who is using rfc2833 DTMF with a non-asterisk SIP provider try inserting this in their dial plan to see if it works (trying to see if the DTMF tones are heard on the PSTN side of the equation). What happens to me is the first digit gets heard but then silence as the next 8 digits are sent. extensions.conf: exten = 55,1,Dial(SIP/sip_provider/55,20,D(123456789)) For your info - here is what I have in my sip.conf: [general] disallow = all allow=ulaw port = 5060 context = incoming maxexpirey=180 defaultexpirey=160 canreinvite=no srvlookup=yes videosupport=no nat=no tos=reliability dtmfmode=rfc2833 [sip_provider] type=friend username=123456789 secret=password host=10.0.0.1 disallow=all allow=ulaw maxexpirey=15 relaxdtmf=yes dtmfmode=rfc2833 nat=no insecure=very canreinvite=no promiscredir=yes Thanks in advance for your help! ___ --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] DTMF doesn't work between Asterisk and Cisco SIP Proxy
Jason, I am at least having similar issues with rfc2833 DTMF: http://bugs.digium.com/view.php?id=10058 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote: Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --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] chan_features.so / asterisk 1.4.5
Joshua Colp schrieb: Jack wrote: Thanks for your answer and sorry for my late response. So what does this exactly mean to me? Can I keep chan_features.so from 1.4.4? What consequences does it have when chan_features.so is disabled und why has this been done? Is chan_features.so related to features.conf? chan_features.so doesn't provide anything useful, it's not used anywhere because it's not really finished. As for why it was changed to be disabled by default oej probably thought that since we aren't using it why do we have it enabled by default. It's fine to live in a world without chan_features.so :) Thank you very much. ___ --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] Transfer Call to Cell Phone
Polycom Phones 1. New call 2. Press 9 access outside line 3. Dial Cell Number 4. Transfer the call that way. Once you initiate a new call you will tie up the second line. Your asterisk box will now be bridging the two lines. The lines will stay tied up until the salesman drops the call. One method you might be able to employ here would be to add a call transfer to the pots lines. Then you would need to send a hook flash to the pots line, and dial the salesman's number when you get the dial tone. Then, depending on how your local Telco supports the call transfer feature, you may be able to free up the line. Not all Telcos support this the same way as some consider it a method of toll avoidance and thus drop the call. This would be possible in an area where a call from party A to party B is a local call and the call from party B to party C is a local call but a call from party A to party C would be a toll call. Since the call from party A to party C is a toll call, the Telco may opt to drop the call. If the transfer part works, there may even be a way to setup the dial plan to intercept your phones call transfer feature and use a 1-2 digit code to select which phone number to send out. I have not done this but I think it is reasonable as I've heard of home users doing it. By the second option, are you talking about the TDMA/GSM gateway? If so, yes this is pretty slick. We considered it initially as well. Our decision not to use it was based on the fact that many of our agents are on different mobile plans. I think when we requested the info from the agents we had 6 different wireless companies represented. Since Sprint/Nextel, Cingular/ATT, T-Mobile don't like to play nicely with other, we didn't expect to see any real savings from the free mobile to mobile calls. This was mainly due to the fact that we don't pay for the agents phones and thus we can't really tell the agents which carriers to use. I do know of a couple of installations where the company does provide the phones and I understand the savings can be significant. I was told by friend that the box they installed paid for itself in just a couple of months. But their phone were already on the same plan. John ___ --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] Inexpensive Layer 3 Switch?
Sorry, I forgot to mention that I want to route between VLANs without an external router and do some simple ACLs to allow PCs on the data VLANs to access the web interface of the Trixbox on the voice VLAN. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Tuesday, June 26, 2007 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inexpensive Layer 3 Switch? You do not need an L3 switch for this, just any managed switch which does vlans Unless there is something else? On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I’d be glad to hear them…so far I’m looking at the SFE2000 from Linksys. thanks No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.9/870 - Release Date: 6/26/2007 10:07 AM ___ --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] More FAX over T1
This is a follow up to an earlier post. Looking for a means to individualize incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to enter the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into the Asterisk box. There is a separate bank of 4 POTS lines going into a FAX server. Looking for a way to assign numbers as incoming FAX lines and have them received with the incoming number intact. Having these forwarded to one of the analog numbers is a thought, but I am concerned about various issues, data corruption, etc, going that route. Thoughts vary to second T1, with channel bank, breaking out some DS0's into a channel bank, or finding a T1/fax board (do they exist?), to go directly into the FAX server (PC/linux based) joe a. ___ --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] AstPligg
Niothing, it just serves a different purpouse. A site like voip-info, astrecipes, asterisk book wiki is for adding content; a site like digg is more for pointing out things you find on the various sites and to share them with other people. l. In data Tue, 26 Jun 2007 15:12:38 +0200, Jon Weisman [EMAIL PROTECTED] ha scritto: Whats wrong w/ voip-info.org? Jon Weisman | Sales Engineer International Bell Communications www.ibell.net - Original Message - From: Alex Robar To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, June 26, 2007 8:38 AM Subject: Re: [asterisk-users] AstPligg Give the new site a break. I think it's a good idea. Sure there are lots of news sites for VoIP, but many of them are poorly designed, and I can't recall any that are very good at letting the users provide the news content. I agree that the name could be better, but after having just tried it out, I really like AstPligg. AR On 6/25/07, Mark Phillips [EMAIL PROTECTED] wrote: Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] -- ___ --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 -- 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] Bridging two PSTN calls
Doug Zingel wrote on Tuesday, June 26, 2007 4:39 AM I'm wondering if its possible to receive a call from an external number (PSTN) say A. Then make a call to another external number (PSTN) say B - and then bridge the two calls so that A is talking to B? Yes, look at blind transfer or attended transfer. What hardware will I need to be able to do this. The same hardware as to set up a system to take calls. The hardware will depend on what interface you will be using to the outside world. IP (SIP)? POTS? DID/DOD trunks over T1? Secondly, if I had x number of simultaneous calls (A talking to B) - how many PSTN lines would I need? I think 2x. That is correct. Don Pobanz ___ --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] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: Sorry, I forgot to mention that I want to route between VLANs without an external router and do some simple ACLs to allow PCs on the data VLANs to access the web interface of the Trixbox on the voice VLAN. thanks The only reason to route the voice VLAN is if you need the phones to access the Internet and/or vice-versa. If you only need to worry about the computers on the data VLAN accessing Trixbox's web interface, I would suggest using the Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on Trixbox, and have the default vlan for the port on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and the phones. The other option is to put in another NIC, one for the voice VLAN, the other for the data VLAN. I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). They're running around $400 right now. If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. But then again, I got a bunch that don't work that I could sell you ;) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Records s as dst
Does doing it this way give you the correct DST?
[macro-dialout]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _X.,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = _X.,n,NoOp(CID_NAME : ${CID_NAME})
exten = _X.,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = _X.,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = _X.,n,NoOp(TRUNK : ${TRUNK})
[rest of your macro]
Jaswinder Singh wrote:
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = s,n,NoOp(CID_NAME : ${CID_NAME})
exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = s,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = s,n,NoOp(TRUNK : ${TRUNK})
exten = s,n,Set(CALLERID(name)=${CID_NAME})
exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
exten =
s,n,Set(PRESENTATION=${IF($[${CID_CLIR}=1]?prohib_not_screened:allowed_not_screened)})
exten = s,n,SetCallerPres(${PRESENTATION})
exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface
exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Playtones(busy)
exten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
Brisbane (07) 3018 2840
Fax (07) 3105 5987
Disclaimer - This email and any files transmitted with it are
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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy
This is usually a Cisco issue. You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs. tracinet wrote: Jason, I am at least having similar issues with rfc2833 DTMF: http://bugs.digium.com/view.php?id=10058 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote: Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --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 ___ --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] SpectraLink SVP protocol support in asterisk
Does anyone know if Asterisk can natively support the SVP protocol from SpectraLink? Thanks, MD ___ --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] Problems with ChanIsAvail always return status 0
Jared: As you see i have the s option. That works fine on Version 1.2. Let me see config the call limit con sip channels it works. Thanks. On 6/25/07, Jared Smith [EMAIL PROTECTED] wrote: On 6/25/07, Alvaro Parres [EMAIL PROTECTED] wrote: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. This isn't really a problem with ChanIsAvail... it's more of a misunderstanding of what's going on. In your case, it appears that your SIP device will accept multiple calls at the same time from Asterisk. So even if your phone is on a call, Asterisk will come along, try to make another call to it, and the phone says Hey, go ahead! I don't mind! You've got quite a few options to solve your problem. While none of them are exactly perfect, it's good to have lots of options: o Try using the 's' option to ChanIsAvail(). (You might have to turn on call limits in sip.conf to get this to work correctly. Last time I played with this, it seems that the limitonpeers setting had to be set to yes as well.) o Use the GROUP() dialplan function to assign calls to call groups, and then use the GROUP_COUNT() function to check to see if that phone is already on any calls. o Turn off call waiting on your IP phone, so that it'll only accept one call at a time o Simply get call limits in sip.conf working correctly. (This is probably the hardest to do, unfortunately.) Hopefully, one of those options will help you out. (I've placed them in the order I'd try... but your mileage may vary.) -Jared ___ --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 -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ --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] DTMF doesn't work between Asterisk and Cisco SIP Proxy
To configure the Cisco for RFC 2833 add the following line to the desired dial-peer dtmf-relay rtp-nte Hope this helps. Ed Nuñez -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, June 26, 2007 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy This is usually a Cisco issue. You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs. tracinet wrote: Jason, I am at least having similar issues with rfc2833 DTMF: http://bugs.digium.com/view.php?id=10058 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote: Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --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 ___ --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] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: This is a follow up to an earlier post. Looking for a means to individualize incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to enter the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. With all due respect, this project should be handed over to whomever has authorization to administer the Asterisk box. We can tell you how to do it in Asterisk, but if you can't take our advice, our ability to help you will be severely limited. Now, we have many, many fax machines. We have our incoming through PRI, and then redirect to a channel bank. We have no problems with fax reception. When we used a Sangoma card, we did, but now that we're back on Digium hardware, we've been doing well, thus far. Probably had to do with the echo cancellation, but without infinite time to troubleshoot, we just had to get it working. I would not recommend passing fax data across the PCI bus between cards. I'm probably just superstitious, but I wouldn't do it. But it would be very simple to do with just a Dial statement. Basically, just go out and try it. Your business requirements and what you're allowed to do obviously drive where your decision is going to go. If you get stuck, and can't find answers through Google or the Wiki, then ask this list. But you can't expect us to tell you what's going to work in your business when you aren't empowered to follow our advice. ___ --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] Inexpensive Layer 3 Switch?
Marty Mastera wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I’d be glad to hear them…so far I’m looking at the SFE2000 from Linksys. thanks I have been eying the Dell webmanaged Gigabit switches. They are inexpensive, managed, support port mirroring, VLAN, QoS, and much more. I am probably going to get the 16 or 24 port for testing. The forwarding rates are an often overlooked spec. $202 (currently on sale) for 16 Gigabit ports is almost too good to be true for a managed switch. I am in no way recommending this switch since I have yet to test it but I certainly will give it a try. http://www.dell.com/content/products/productdetails.aspx/pwcnt_2716?c=uscs=04l=ens=bsd 16 10/100/1000 BASE-T ports Auto-negotiation for speed, duplex mode and flow control Auto MDI/MDIX mode and flow control Integrated Port LEDs Individual port controls Performance Switching Capacity 32.0 Gbps Forwarding Rate 23.7 Mpps Management Web-based management interface BootP/DHCP IP address management or Static IP address assignment RMON statistics Class of Service Four priority queues per port Adjustable WRR and strict priority Layer 2 IEEE 802.1p tagging and port-based priority Layer 3-aware prioritization using DSCP values Security Switch access password protection (read-only and read-write access) Restricted IP address VLAN IEEE 802.1Q port-based tagging up to 64 VLANs Honors all 4096 VLAN tags Switching Features Link Aggregation, up to six groups and up to four aggregated links per group (IEEE 802.3ad) Port mirroring (up to four source ports) Jumbo frame support up to 9000 Bytes (2716 2724 only) Availability Firmware Uploads to the Switch Broadcast Storm Control Virtual Cable Tester by Marvell^® Optical Transceiver Diagnostics Chassis *Dimensions:* 1.70 in (H) x 12.99 in (W) x 9.07 in (D) *Height:* 1U rack *Weight:* 6.16 lbs *Voltage:* 100-240VAC, 50-60Hz Standards Supported IEEE802.3 CSMA/CD IEEE802.3u 100BaseTx IEEE802.3z/ab 1000BaseT IEEE802.3x Flow Control IEEE 802.1p Environmental Operating Temperature: 0º C to 45º C (32º F to 113º F) Storage temperature: -20º C to 70º C (-4º F to 158º F) Operating Humidity: 10% to 90% Relative Humidity Storage Humidity: 10% to 95% Relate Humidity Power Maximum Power: 1.0A @ 100V Once upon a time I would say Go Cisco, but I think Dell has come a long way with their servers and switches. Thanks, Steve Totaro ___ --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] Ring/Off-hook in strange state 6
Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --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] Inexpensive Layer 3 Switch?
If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. What model did you use? I've been very happy with the FS728TP as I mentioned earlier. I haven't had any problems so far. Granted I haven't had to call Netgear so I don't have anything on which to judge their service. I will say that unless your dealing with a very small system, you should probably steer away from the FS726TP. It only supports PoE on the first 12 ports and doesn't have anywhere near the features of the FS728TP. John ___ --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] More FAX over T1
. . . With all due respect, this project should be handed over to whomever has authorization to administer the Asterisk box. We can tell you how to do it in Asterisk, but if you can't take our advice, our ability to help you will be severely limited. Thanks. Point taken. I'm, unfortunately, playing a form of monkey in the middle. Seems the Vendor is unwilling to, unable to, or is outrageously priced. I am not privy to any of those discussions. I view this as a learning experience for me. Now, we have many, many fax machines. We have our incoming through PRI, and then redirect to a channel bank. We have no problems with fax reception. When we used a Sangoma card, we did, but now that we're back on Digium hardware, we've been doing well, thus far. Probably had to do with the echo cancellation, but without infinite time to troubleshoot, we just had to get it working. This install uses a Sangoma card. Could you expand on redirect to a channel bank? Could you illuminate the connectivity for me? A single T1 connects to??? Is the Digium card smart as in, can it break out DS0 line(s) on a second port (to go to the channel bank)? I am not that familiar with that technology. As may be evident. joe a. ___ --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] Slip Events
All, I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy errors. I rewired this thing three times, then I connected the same cable from the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events and frame sync errors went away, so the cable is good. STX -- DSX Panel --Asterisk I've confrimed all wiring is good, not sure what the issue is, tried playing with the clock source as well, but no dice. Any ideas? Errors: CRC Errors Frame Sync Errors Slip Events 1 Error every second Zaptel.conf loadzone= us defaultzone = us span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 bchan=73-95 dchan=96 TIA, Jon Weisman | Sales Engineer International Bell Communications www.ibell.net www.ibellhost.com www.aeronhelpdesk.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] Ring/Off-hook in strange state 6
I have seen this on cards waiting for the callerID and there being a problem with the callerid signal. Is callerid working on theses lines? On 6/26/07, Alex Mcdowell [EMAIL PROTECTED] wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.clike someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --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 -- Bruce Reeves Nortex Networks ___ --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] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: Thoughts vary to second T1, with channel bank, breaking out some DS0's into a channel bank, or finding a T1/fax board (do they exist?), to go directly into the FAX server (PC/linux based) It looks to me like you have two choices. The first you probably can't do. That is, get a two port board in the Asterisk system with the second T1 going into an Eicon Board in a Hylafax system. Then, you can assign DIDs with whatever web interface you have on this Asterisk system to go to the second T1 port. The second alternative is to get a second T1 and an Eicon card going into a Hylafax Server. This solution has a big monthly expense to it, especially if you aren't fully utilizing all channels on your existing T1. I'm moving towards the first solution being we send out (and receive) large faxes and the IAXModem solution, because of patent issues, is not able to send at the faster speeds. I've received complaints about our slow fax machine. The Eicon card can support the faster transmission. ___ --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] Bad Echo between SIP calls
First of all, Alex, sorry for not seeing your reply. Nearly two weeks ago now :( Honestly, with canreinvite=yes, I'm not sure what is meant by the signalling still travels through asterisk... I would ASSUME that includes out-of-band dtmf as well. Sorry! Moj Alex Crow wrote: Moj, Does this mean that even out-of-band DTMF still gets sent SIP-phone--SIP-phone without Asterisk hearing them? (eg RFC DTMF, can't remember the number right now) Forgive me for butting into this thread but this is interesting... Cheers Alex On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan Company, LLC wrote: theoretically, with canreinvite=yes, it's phone - phone. with canreinvite=no, it's phone - asterisk - phone. BUT there are a few reasons which canreinvite=yes will not be this way. If for example you have a T or a t in the Dial string, asterisk will _remain_ in the media path so it can still detect the DTMF requests for transfer. Moj Deepak Naidu wrote: Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak */Matthew Fredrickson [EMAIL PROTECTED]/* wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --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 Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU. ___ --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 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] Ring/Off-hook in strange state 6
Daniel already pointed you in the right direction. I have seen this error many times, but it never causes a problem. Alex Mcdowell wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy
Buddies, Thanks for you response. I have resolved the issue,it was not the DTMF mismatch between Asterisk and Cisco proxy. In fact,there is a Convedia media box behind Cisco proxy as conference bridge,after checked the whole trace through the patch,I found that my asterisk send video codec information in the SDP of invite,but the Convedia media box doesn't support video,it got the request but did not reject,just add a blank IP address 0.0.0.0 in SDP of 200 OK,so there were two sections in 200 OK SDP,one is audio section with audio IP address and port,the other is video section with a 0.0.0.0 IP address. When Asterisk got 200 OK,it was strange that it treated the video IP 0.0.0.0 as audio address,so the call was established but could not go on,that was why I input anything it did not work,I think. When I disabled the video support in Asterisk,it worked. On 6/26/07, Ed Nuñez [EMAIL PROTECTED] wrote: To configure the Cisco for RFC 2833 add the following line to the desired dial-peer dtmf-relay rtp-nte Hope this helps. Ed Nuñez -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, June 26, 2007 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy This is usually a Cisco issue. You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs. tracinet wrote: Jason, I am at least having similar issues with rfc2833 DTMF: http://bugs.digium.com/view.php?id=10058 On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote: Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-liteAsterisk---Cisco SIP proxySIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in the log of asterisk,I can find that asterisk recognized all the digitsI tried rfc2833,inband,info in the dtmfmode parameter,but did not work ,I'm not sure whether asterisk send the right dtmf to cisco proxy,how can I track that? I made another test,dialing from x-lite registered with Cisco proxy to voicemail service of Asterisk. x-liteCisco SIP proxyAsterisk---Voicemail service Both the call and dtmf worked fine,I can input my mailbox number and password and listen my voicemail.both rfc2933 and inband worked in this situation,but not info. My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in the section of xlite and the trunk to cisco proxy,just configure the dtmfmode in sip.conf. When I used rfc2833,I can see the log in asterisk as : [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on SIP/-08269470 [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on SIP/-08269470, duration 160 ms [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on SIP/-08269470 [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on SIP/-08269470, duration 140 ms and when I used inband,I can see : [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on SIP/-09d916c0, duration 0 ms [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on SIP/-09d916c0, duration 0 ms Is that right?Can I check what digits that asterisk sent out ? How can I track where is wrong with the dtmf?Did asterisk send dtmf to Cisco proxy correctly? I really have no idea about that.Please advise.Thank you very much ___ --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 ___ --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
Re: [asterisk-users] Inexpensive Layer 3 Switch?
The only reason to route the voice VLAN is if you need the phones to access the Internet and/or vice-versa. If you only need to worry about the computers on the data VLAN accessing Trixbox's web interface, I would suggest using the Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on Trixbox, and have the default vlan for the port on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and the phones. The other option is to put in another NIC, one for the voice VLAN, the other for the data VLAN. I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). They're running around $400 right now. If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. But then again, I got a bunch that don't work that I could sell you ;) Ahh, interesting idea…if I understood correctly, you’re basically using a layer 2 switch and trunking the voice and data VLAN to the asterisk box and doing the routing and ACL work there? Advantage is lower cost because you don’t need a layer 3 switch anymore and don’t have to learn a new CLI or other config method.? Here’s a bit more information…the client is a building owner who occupies the first floor and is renting out the rest of the building. In addition to his own voice/data network (which would be on separate VLANs) they want to offer the building tenants the ability to use their PBX and internet connection. Due to a quirk in the service providers SIP ALG all IP phones in the building must be on the same network (VLAN) which I don’t see a problem with, but each tenant’s data will be in a separate VLAN. I’m thinking I could trunk the voice VLAN and all of the individual tenant data VLANs to the Trixbox to allow them access to the web interface? Any other ideas out there based on this scenario? Thanks again No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.9/870 - Release Date: 6/26/2007 10:07 AM ___ --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] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
This install uses a Sangoma card.
Could you expand on redirect to a channel bank? Could you illuminate
the connectivity for me?
A single T1 connects to??? Is the Digium card smart as in, can it
break out DS0 line(s) on a second port (to go to the channel bank)?
What we did is have 1 PRI (over T1 in the US) coming in from the telco, into
a 4-port T1 card. Then, we have 2 channel banks coming off of it: 1 has an
external echo can, and goes to our in-house phone extensions. Cordless
phones, wall-hanging phones, and anywhere that we couldn't get 2 pair into.
Basically any required analog that a person would be on. The second channel
bank has no echo can on it, and connects to our modems, fax machines, etc.
Assume for a moment that your incoming lines are in zaptel group 0, your
voice channel bank lines are in group 1, and your other channel bank is in
group 2
exten = 55,1,Goto(default,1000,1) ;go into the internal context to
route the call
exten = 56,1,Dial(Zap/25) ;ring one phone
exten = 57,1,Dial(Zap/G2/${EXTEN}) ;go out group 2, starting at
highest channel number, since the incoming calls probably start at the
lowest channel numbers, and best not to have any contention, in my opinion
In this way, Asterisk will establish a new call going out Group 2 and dial
your number. The box receiving the faxes will get the extension, and as long
as you've left your DID in there, that's what will get passed. So, it will
appear to your fax box as if it were sitting on the PSTN. Asterisk just has
to know which DIDs should have the calls passed along.
Now, in practice, we do a lot more than the above snippet, and use macros
extensively. But this should get you pushed in the right direction.
Hope that helps,
David
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Re: [asterisk-users] More FAX over T1
. . . It looks to me like you have two choices. The first you probably can't do. That is, get a two port board in the Asterisk system with the second T1 going into an Eicon Board in a Hylafax system. Then, you can assign DIDs with whatever web interface you have on this Asterisk system to go to the second T1 port. The second alternative is to get a second T1 and an Eicon card going into a Hylafax Server. This solution has a big monthly expense to it, especially if you aren't fully utilizing all channels on your existing T1. I'm moving towards the first solution being we send out (and receive) large faxes and the IAXModem solution, because of patent issues, is not able to send at the faster speeds. I've received complaints about our slow fax machine. The Eicon card can support the faster transmission. No solution, thus far, seems very cost effective for this client. The Eicon and other T1/fax board are in the 4-7K$ range. As this venture is still in it's infancy, it would not be acceptable to shell out such, at the moment. One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. joe a. ___ --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] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: The only reason to route the voice VLAN is if you need the phones to access the Internet and/or vice-versa. If you only need to worry about the computers on the data VLAN accessing Trixbox's web interface, I would suggest using the Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on Trixbox, and have the default vlan for the port on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and the phones. The other option is to put in another NIC, one for the voice VLAN, the other for the data VLAN. I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). They're running around $400 right now. If you really need layer3 support, I would steer clear of the Netgear. I've had a lot of problems with them, and the support was disappointing. But then again, I got a bunch that don't work that I could sell you ;) Ahh, interesting idea…if I understood correctly, you're basically using a layer 2 switch and trunking the voice and data VLAN to the asterisk box and doing the routing and ACL work there? Advantage is lower cost because you don't need a layer 3 switch anymore and don't have to learn a new CLI or other config method.? Here's a bit more information…the client is a building owner who occupies the first floor and is renting out the rest of the building. In addition to his own voice/data network (which would be on separate VLANs) they want to offer the building tenants the ability to use their PBX and internet connection. Due to a quirk in the service providers SIP ALG all IP phones in the building must be on the same network (VLAN) which I don't see a problem with, but each tenant's data will be in a separate VLAN. I'm thinking I could trunk the voice VLAN and all of the individual tenant data VLANs to the Trixbox to allow them access to the web interface? Any other ideas out there based on this scenario? We do something somewhat similar. Each switch has 2 data VLANs, and also is part of the Voice VLAN. Each VLAN for data is routed, but the voice VLAN only carries voice traffic. Our Asterisk server does not route packets between the networks. So, aside from some nasty attacks that sniff and replicate VLAN headers, our voice network is pretty secure. So our network has 20 different data VLANs (again, 2 per edge switch), 1 server VLAN, 1 voice VLAN, 1 wireless VLAN, and one DMZ VLAN. The data and server VLANs are all routed, and everything else is not. They have to go through some type of bridge between the networks. For wireless, that's our wireless switch. For the DMZ, it's our firewall. The voice VLAN can only reach our Asterisk box. If you use a SIP provider, you may have to either take another approach, or realize that all SIP traffic will have to remain on the host (i.e. reinvites are bad when you don't have a network path from A to B). But we're strictly IAX between offices, and PSTN thru PRI. ___ --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] kore dump
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it's running, no harm done, and if it crashes, the cron job will make sure that it's started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --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] TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience
Has anyone successfully run * 1.4 with the following configuration (or something very similar)? HP DL380 G5 (3Ghz Xeon) CentOS 4.5 (kernel 2.6.9-55) Asterisk 1.4.5 (or 1.4.4) Zaptel 1.4.3 (or 1.4.2.1) TE412P TDM400B (2x FXO and 2x FXS modules) I've had this rig running * 1.2.18 with Zaptel 1.2.17.1 for several months without any issues. Upon trying to upgrade to * 1.4.4 and Zaptel 1.4.2.1 a few weeks ago, I saw several kernel panics, easily reproduceable with a load test suite that bridged calls from the PSTN to SIP desk phones. The one time I exposed it to real world traffic (a small office, 30 extensions), I saw five panics in the space of three hours. Interestingly, when the load test only walked through the IVR and queues, I couldn't get it to panic, even if I filled the TE412P and had 90 simultaneous calls going through the system. Only bridged calls seem to cause problems. The panics didn't seem to follow any particular pattern. I saw NMI traps a few times, then failures in the wct4xxx interrupt handler, a few swap tainted errors, etc. To rule out an interaction between the TE412P and the HP motherboard, I moved the installation to an Intel server board (model SE7320) with a 3 Ghz Pentium D. The hard drive was a direct clone from the HP. Running the same load test against 1.4.4/1.4.2.1 - no panics. Running the load test against 1.4.5/1.4.3 - no panics. I'm now nearly 6 hours into having the system exposed to real-world traffic - no panics. At this point, I'm pretty much ready to just put this Intel board into a rackmount server and be done with it, but I am interested to see if anyone else has seen similar problems, or if anyone has run this configuration without any problems. Part of me is saying I can't be the only person to run * 1.4 on a current HP server, which in turn leads me to wonder if this is an incompatibility with the specific board I have, or if I've got a faulty server (though I have run the full HP diagnostic suite several times). Thanks for any feedback you can provide. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. Sure! You should already have the required script. Just run it from safe_asterisk. Here is a link with more info: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs -html/x389.html John ___ --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] kore dump
use the safe_asterisk script it will restart asterisk if it crashes and it enables core dumps (your core size limit is probably set to 0 when you start asterisk). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Tuesday, June 26, 2007 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [asterisk-users] kore dump I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it's running, no harm done, and if it crashes, the cron job will make sure that it's started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --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] Transfer Call to Cell Phone
Well seems like I am already doing first method minus the extension. We do have full features on our lines so both lines are free once the transfer is complete. We also have toll calls on our lines so it would not be a problem, so I do not have to worry about ATT dropping the call. I tried to create a separate extension for this but it did not work. The only way I can get it to work is by have the call on the 1st line then transfer it out on the 2nd line. After that is complete both lines are free. A worst case scenario would be where our sales volume picked and we needed to transfer a call and could not because of the slots are filled. Can you give an example of creating an extension which points to a cell phone. Secondly how can you have if no one answers an extension it dials the cell number next. That maybe answered in the example. I have the system setup so it just dials out which ever line is not busy. Thanks! Otis John Faubion wrote: Polycom Phones 1. New call 2. Press 9 access outside line 3. Dial Cell Number 4. Transfer the call that way. Once you initiate a new call you will tie up the second line. Your asterisk box will now be bridging the two lines. The lines will stay tied up until the salesman drops the call. One method you might be able to employ here would be to add a call transfer to the pots lines. Then you would need to send a hook flash to the pots line, and dial the salesman's number when you get the dial tone. Then, depending on how your local Telco supports the call transfer feature, you may be able to free up the line. Not all Telcos support this the same way as some consider it a method of toll avoidance and thus drop the call. This would be possible in an area where a call from party A to party B is a local call and the call from party B to party C is a local call but a call from party A to party C would be a toll call. Since the call from party A to party C is a toll call, the Telco may opt to drop the call. If the transfer part works, there may even be a way to setup the dial plan to intercept your phones call transfer feature and use a 1-2 digit code to select which phone number to send out. I have not done this but I think it is reasonable as I've heard of home users doing it. By the second option, are you talking about the TDMA/GSM gateway? If so, yes this is pretty slick. We considered it initially as well. Our decision not to use it was based on the fact that many of our agents are on different mobile plans. I think when we requested the info from the agents we had 6 different wireless companies represented. Since Sprint/Nextel, Cingular/ATT, T-Mobile don't like to play nicely with other, we didn't expect to see any real savings from the free mobile to mobile calls. This was mainly due to the fact that we don't pay for the agents phones and thus we can't really tell the agents which carriers to use. I do know of a couple of installations where the company does provide the phones and I understand the savings can be significant. I was told by friend that the box they installed paid for itself in just a couple of months. But their phone were already on the same plan. John ___ --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] kore dump
On 6/26/07, Ed Nuñez [EMAIL PROTECTED] wrote: I have not been able to locate where the core dump file is being saved. I can't find it in my TMP directory. Check the directory in which you're starting Asterisk. It doesn't sound like you're using the Red Hat initscript to start Asterisk, so you're probably starting it manually. I'd check there first. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it's running, no harm done, and if it crashes, the cron job will make sure that it's started every 60 seconds. Check out the safe_asterisk script that's installed with Asterisk. Again, I recommend you check out the initscript, which calls safe_asterisk and restarts Asterisk if it crashes. It also ensures that your core dumps end up in /tmp where you can find them more easily. On Red Hat and CentOS, you can install the initscript by typing make config in the Asterisk source directory. -Jared ___ --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] More FAX over T1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. That's definitely an idea. If you don't need the Caller ID on the fax (and in most cases, you probably don't), this might be your best solution. Assuming, of course, the faxmodems on Hylafax are picking up the caller ID and you have Caller ID from the phone company. That would take up 2 of your PRI channels, though, per fax reception. ___ --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] More FAX over T1
On 6/26/2007 at 3:04 PM, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. That's definitely an idea. Wow, you mean I actually had one??? g If you don't need the Caller ID on the fax (and in most cases, you probably don't), If you mean (not) printed on the FAX, yes (no ?) it is *not* needed. this might be your best solution. Assuming, of course, the faxmodems on Hylafax are picking up the caller ID and you have Caller ID from the phone company. Worthy of investigation. That would take up 2 of your PRI channels, though, per fax reception. That should not be a problem, at this point. joe a. ___ --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] kore dump
Ed, I am having a problem with Asterisk frequently crashing on me as well. I just run it under supervise: http://cr.yp.to/daemontools/supervise.html This way it will be restarted if svc determines it isn't running. Eric On Tue, 2007-06-26 at 13:22 -0500, Ed Nuñez wrote: I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what’s causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can’t find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If it’s running, no harm done, and if it crashes, the cron job will make sure that it’s started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --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 -- Eric Lubow LinkExperts, Inc. Systems Administrator e: [EMAIL PROTECTED] w: www.linkexperts.com p: 212.542.5201 ___ --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] Cisco 7941 localized menus with SIP firmware
Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring/Off-hook in strange state 6
I don't have caller ID at all, not on the verizon side and usecallerid=no in zapata.conf. I do, however have the DSL on this line. I have a splitter and then I have a filter on the asterisk side. I am guessing this is the root of the problem. Thanks for any insight.-Alex On 6/26/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Daniel already pointed you in the right direction. I have seen this error many times, but it never causes a problem. Alex Mcdowell wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote: Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Has anyone written a configuration tool for Cisco's that generates the correct XML files? Wouldn't that be a useful thing? Steve -- 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] Fax Throughput
Matt, thanks for pointing me to zaptel.conf; I'm new to Asterisk and can use all the help I get! Here are the non-comment lines from zaptel.conf (not set up by me): span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 loadzone = us defaultzone=us The first span is connected to the PSTN. The second is connected to a Windows-based server using Dialogic hardware and custom software. The second span has a clock priority equal to the first one. I'm guessing that this has the effect of ignoring clock from the first span (same as '0') and using clock from the second. Not good. I've changed the clock priority for span 2 to '0'; if we lose the PSTN we'll rely on the Digium card for clock. Fax throughput seems fine with this change. In zapata.conf I find: ; Network Side signalling = pri_cpe group = 1 context = pstn-inbound channel = 1-23 ; IVR Side signalling= pri_net group = 2 context = ivr-inbound channel = 25-47 The default would be switchtype=national, which is correct. I see that for 'signalling', 'group' and 'context' = has been used, rather than the = that I see in documentation. Does this matter? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 26, 2007 9:22 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Throughput Can you post your zaptel.conf so we can verify your timing settings? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 25, 2007, at 11:10 PM, Don Kelly wrote: I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. Any suggestions? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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] More FAX over T1
On 6/26/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote: One idea is to utilize DID, and have Asterisk forward the calls to the current FAX lines, preserving the DID as Caller ID. I am fairly sure Asterisk itself can do this. (The call would appear to be from this assigned ID). If so, I could, apparently, massage Hylafax into dealing with each FAX based on the Caller ID. That's definitely an idea. If you don't need the Caller ID on the fax (and in most cases, you probably don't), this might be your best solution. Assuming, of course, the faxmodems on Hylafax are picking up the caller ID and you have Caller ID from the phone company. That would take up 2 of your PRI channels, though, per fax reception. I think I read that you have 4 fax lines. If this is correct, then the calculation is thus: 4 lines * 30 per month = 120 per month. channel bank = $200 new (for faxes, I have never had a problem with zhone CBs found at http://www.digital-loggers.com/CB.html) 2-port T1 card = $900 new, for a total of $1100 in equipment (a one-time cost) So, if you keep the solution going for more than 9 months, you'll come out ahead just buying the equipment. If you pay more than $30 per month per phone line, your break-even will be much quicker. Also, if you ever needed to add more lines, you already can have 24 faxes through Asterisk, and your fax server would be the bottleneck. This is why we run all of our fax lines off of our PRI, even though our local dialtone provider tried to convince us to buy POTS lines for each one. At around $30 per line (not including taxes), it just doesn't add up. ___ --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] Cisco 7941 localized menus with SIP firmware
2007/6/26, Steve Kennedy [EMAIL PROTECTED]: On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote: Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Has anyone written a configuration tool for Cisco's that generates the correct XML files? Wouldn't that be a useful thing? Steve -- 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 Hi, Do you mean all 7941 menus provisionning is about, is downloading a given XML file ? Is it easy to get a sample of this file ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
Vadim Berezniker wrote:
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your
core size limit is probably set to 0 when you start asterisk).
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed Nuñez
*Sent:* Tuesday, June 26, 2007 2:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
*Subject:* [asterisk-users] kore dump
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.
My PBX has experienced several core dumps the last couple of days and
I am not sure if this is what’s causing it, but it always seems to
happen when a particular extension on a grandstream phone uses ChanSpy
SIP group.
I have not been able to locate where the core dump file is being
saved. I can’t find it in my TMP directory.
I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump. I was thinking of setting up a cron job
to launch Asterisk every minute. If it’s running, no harm done, and if
it crashes, the cron job will make sure that it’s started every 60
seconds.
Any suggestions?
Thank you
Ed Nuñez
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If that fails you could always try something like:
*/2 * * * * /bin/ps -C /usr/bin/asterisk || { /usr/bin/asterisk }
or so...
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g'
Wise men talk because they have something to say;
fools, because they have to say something. -- Plato
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] zaptel 1.2.18 and HPEC
Paul Hales wrote: Any idea why I can't build HPEC for zaptel 1.2.18? It builds fine with 1.4.3... I don't know why this change just happened, but it has been fixed in revision 2668 of SVN branch-1.2 of Zaptel. You can fix it on your system by adding the following line: .EXPORT_ALL_VARIABLES: just above the 'FORCE:' line in the Makefile. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] No such host error from SIP for non-peer configuration.
Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten = 500,n,Dial(SIP/[EMAIL PROTECTED]) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in sip.conf and have to use this peer in extension.conf exten = 500,n,Dial(SIP/[EMAIL PROTECTED]) Is there any configuration in sip.conf restrict me to do this? 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] Bridging two PSTN calls
On 6/26/07, Doug Zingel [EMAIL PROTECTED] wrote: I'm wondering if its possible to receive a call from an external number (PSTN) say A. Then make a call to another external number (PSTN) say B - and then bridge the two calls so that A is talking to B? What hardware will I need to be able to do this. Yes, that's one of the many things Asterisk can do. Obviously, you'd need a hardware card to connect Asterisk to the phone lines. You'd need FXO ports to connect Asterisk to analog phone *lines*, and FXS ports to connect Asterisk to analog *phones*. You could also use IP phones, and/or a VoIP provider. Secondly, if I had x number of simultaneous calls (A talking to B) - how many PSTN lines would I need? I think 2x. I know this sounds ridiculously inefficient - but I think the experiment might open some avenues. Yes, you would need 2x the number of calls for a straight analog setup, unless you try to get fancy and setup flash transfers with your telco. Again, you could use a VoIP provider to get around the hardware costs of analog ports if port cost is a concern to you. -Jared ___ --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] No such host error from SIP for non-peer configuration.
I figured it out. srvlookup=no On 6/26/07, Lucian Romi [EMAIL PROTECTED] wrote: Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten = 500,n,Dial(SIP/[EMAIL PROTECTED]) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in sip.conf and have to use this peer in extension.conf exten = 500,n,Dial(SIP/[EMAIL PROTECTED]) Is there any configuration in sip.conf restrict me to do this? 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] Provisioning Linksys WIP330 phones
http://www.abptech.com/support/qa/index.php?target=linksy_remote_p _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Sunday, June 24, 2007 10:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Provisioning Linksys WIP330 phones We're contemplating Linksys WIP 330 phones, but we're concerned about configuration effort. Does anyone have the file format for an XML file to configure this phone? We got auto-provisioning to D/L a file, but the XML file format seems to be a secret... Thanks, Michelle ___ --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] kore dump
I am having a problem with Asterisk frequently crashing on me as well. I just run it under supervise: But that's just a band-aid. If it crashes, it takes all calls with it. Hardly a good thing, unless you only have 1 call at a time -- then it's probably no the end of the world. I still don't know what's up with the crashes but here are two observations I made: 1) I moved the same installation from one hardware to another. On the former hardware it would crash every 2 weeks, on average. On the new hardware, it has not yet crashed and it has 9 weeks of uptime. Same call volume, same devices, same network. I'm running asterisk chroot'ed so all libraries, binaries, config files, etc. are identical. Only the hardware and kernel are different. 2) The same old hardware has been in service for 3 years and no other programs crash on it. Ever. It's no unusual seeing uptime for say qmail, samba or bind of 200+ days. I have therefore reasons to believe that the hardware is OK. So go figure. And BTW, the crashes (based on the core dumps) are always at a different place. There is no consistency. Right now I'm just glad it no longer core dump on me :). --Luki ___ --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] Modification of Caller ID based on context
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250).
This is set in sip.conf for each user. Each user has a local DID
as well (hosted through Vitelity, for example (555)111-). The
problem is that this extension was being passed to the outside
world. I currently have a SetCallerID command changing the
CallerID to our main office number, but some users want their DID
sent, not the general number.
The problem is that if their caller ID is set to their DID, when
users hit redial on their phones internally they dial out and
back in. I corrected this by putting each DID in extensions.conf
under their three digit extension, but that seems a bit like a
kludge obviously.
I'm looking for a method of sending the internal three digit
extension only when a user is dialing another user internally,
otherwise it will send their DID. Is their a method to do this in
the dial plan? Anyone have an example of how to accomplish this?
Thanks in advance.
Mike,
I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 = 555111 ... extdid/250 =
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:
exten = NXXNXX,n,Set(outgoingCID=MAINNUMBER)
exten = NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)
exten = NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})
exten = NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})
...
You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 = 200 ...
didext/5551112272 = 250.
exten = NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)
OR you could do something similar with LOCAL channels or with a Dial
command.
-Matthew
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Re: [asterisk-users] AudioCodes Gateway and Asterisk
When you see [ERROR] in the Message Log, either the MP firmware is buggy or the far end is sending something out of spec in the SIP Message. You'll need to upgrade to the latest MP firmware then report this to whomever you bought it from. Or fix the far end to send the message in spec or form that doesn't cause the [ERROR]. Also, do your supporter a favor and don't paste those logs directly into emails. The wrap makes them horrible to read and they can't send them on to Audiocodes like that. Put them in a text file which preserves the line length. Regards, Shanon http://www.abptech.com/support/qa/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, June 24, 2007 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk - Original Message - From: Shanon Swafford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, June 21, 2007 6:27 PM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk On 6/21/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Are you sure, your asterisk supports G729? It isn't supported by default, you need additional modules or hardware cards for G729 support. If it is - what are you using for G729 - that might help to identify the problem. Regards, Atis If the AudioCodes is sending back that 415, the Message Log in the AudioCodes is invaluable. Set your debug level to 5/6 and watch it while you make test calls. Once you learn how to interpret this output, you'll be well on your way with AudioCodes. If G729 is active on the MP, but still giving back that error, G729 might not be in a profile if you are using them. Also, firmware that comes on the MPs is normally sorta buggy, ask your reseller for the latest version. http://www.abptech.com/support/faqs/ Regards, Shanon ABP Technology Shanon, The audiocodes were preftctly with other providers using G729. It's just having an issue with asterisk. Here is the output from the AudioCodes: Log is Activated 12d:23h:36m:17s ( lgr_flow)(828 ) Incoming SIP Message from XXX.XXX.XX.XXX:5060 [File: Line:-1] 12d:23h:36m:17s INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: lt;sip:XXX.XXX.XX.XXX;ftag=as4a537e63;lr=ongt; Via: SIP/2.0/UDP XXX.XXX.XX.XXX;branch=z9hG4bK1db7.ecac6e86.0 Via: SIP/2.0/UDP XXX.XXX.XX.XXX:5060;branch=z9hG4bK1ef72aa9;rport=5060 From: 55560888 lt;sip:[EMAIL PROTECTED]gt;;tag=as4a537e63 To: lt;sip:[EMAIL PROTECTED]gt; Contact: lt;sip:[EMAIL PROTECTED]:5060gt; Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Enswitch Max-Forwards: 16 Date: Wed, 20 Jun 2007 19:44:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 29170 29170 IN IP4 XXX.XXX.XX.XXX s=session c=IN IP4 XXX.XXX.XX.XXX t=0 0 m=audio 14878 RTP/AVP 18 0 8 10 3 111 5 7 110 97 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no;mode-change-period=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/80 12d:23h:36m:17s ( sip_stack)(830 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(831 ) | | new GetNewSIPCall created - #8 [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(832 ) new AcSIPCallAPI created - #5 [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_mngr)(833 ) Resource StackSession lt;#5gt; Allocated [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(834 ) | |(SIPTU#8)INVITE State:Idle() [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(835 ) SIPCall(#8) changes state from Idle to Invited [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(836 ) AcSIPParser: Problem in AcSIPCallAPI::ParseSDP [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(837 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ' [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(838 ) !! [ERROR] Message type: INVITE [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(839 ) !! [ERROR] Source header: [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(840 ) !! [ERROR] Line: 20. Column: 27 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(841 ) | | | #5:SIP_SETUP_EV([EMAIL PROTECTED]) [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_ses)(842 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_ses)(843 ) lt;SESSION #5gt; SendToCall - event: NEW_CALL_EV m_Call = 32173400 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(844 ) | | #5:NEW_CALL_EV:([EMAIL PROTECTED]) [File: Line:-1] 12d:23h:36m:17s ( lgr_psbrdif)(845 ) AcBoard::GetTrunkGroupId - No entry found for: DstNum:55560918 SrcNum:55560888
[asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working
Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to transmit frame type 1024, while native formats is 4 (read/write = 4/4) Jun 26 17:53:52 WARNING[14248]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (160)? Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received The call continues with this error until I hang up. I have been adjusting the dial-peer dtmf settings in the 2600 and have tried all the dtmf settings in Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Modification of Caller ID based on context
Great examples Matthew, really appreciate it. This is exactly what I've been
searching for!
On 6/26/07, Matthew Brothers [EMAIL PROTECTED] wrote:
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250).
This is set in sip.conf for each user. Each user has a local DID
as well (hosted through Vitelity, for example (555)111-). The
problem is that this extension was being passed to the outside
world. I currently have a SetCallerID command changing the
CallerID to our main office number, but some users want their DID
sent, not the general number.
The problem is that if their caller ID is set to their DID, when
users hit redial on their phones internally they dial out and
back in. I corrected this by putting each DID in extensions.conf
under their three digit extension, but that seems a bit like a
kludge obviously.
I'm looking for a method of sending the internal three digit
extension only when a user is dialing another user internally,
otherwise it will send their DID. Is their a method to do this in
the dial plan? Anyone have an example of how to accomplish this?
Thanks in advance.
Mike,
I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 = 555111 ... extdid/250 =
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:
exten = NXXNXX,n,Set(outgoingCID=MAINNUMBER)
exten = NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)
exten = NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})
exten = NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})
...
You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 = 200 ...
didext/5551112272 = 250.
exten = NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)
OR you could do something similar with LOCAL channels or with a Dial
command.
-Matthew
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Re: [asterisk-users] kore dump
just an idea, but maybe qmail, samba, and bind have a smaller memory footprint than an in-use asterisk? can you take the hardware offline long enough for a memtest? Moj Luki wrote: It's no unusual seeing uptime for say qmail, samba or bind of 200+ days. ___ --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] Ring the second line when 1st line is busy
Do any one any clue.
This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you can
see the extensions asssigned to a user as.
555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
8555 --- Line 2 -- Extensions which registers with SIP(Asterisk) -- User A
So when now someone calls one Extension 555 to User A, scenario as below.
1) If he is busy on the first line Ext 555, then ring Line 2 Ext 8555, if no
response send to voicemail
2) If he doesnt picks the line 1 Ext 555, then send to voicemail rather then
ringging on his second line ie Ext 8555.
This is what I need, if I can dow it with Follow me, then how, if through ring
group how.
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone on
line 1(555) after 10 seconds then ring the line 2(8555). But this is not a
standard telephony which I have been advised to change like below.
If someone calls Ext 555 its busy(means on a phone with someone), then only
ring the second line ie 8555 even if that is busy send on voicemail.
If the first line 555 is free no one picks up then let it go on the
voicemail not second line, bcos now no one has picked the phone nor busy.
I could find anyway to do in FreePBX, so was wondering how about doing this.
Thanx for any input.
exten = 555,1,Macro(exten-vm,555,555)
exten = 555,n,Hangup
exten = 555,hint,SIP/555
exten = ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
--
Deepak
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Re: [asterisk-users] access to asterisk server since internet
From what you provided, I'm not sure that 'firewall is disabled' will help you. Your firewall probably needs to be configured to forward some ports to the asterisk box's internal IP address. I usually do the following: For SSH connections to the box to manage it: forward a.b.d.c's external port 22 to 192.168.1.111's port 22 if using SIP, also: forward a.b.d.c's external port 5060 to 192.168.1.111's port 5060 forward a.b.d.c's external ports 1-2 to 192.168.1.111's ports 1-2. and, if using IAX, also the following: forward a.b.d.c's external port 4569 to 192.168.1.111's port 4569 if you're running some manner of web management GUI, you might: forward a.b.d.c's external port 80 to 192.168.1.111's port 80 BUT be wary that any GUI you choose is NOT NECESSARILY safe to point at the public. Someone could compromise it, who knows. Honestly I would use the SSH port, port 22, to create a tunnel to port 80 on the asterisk box. Moj skalli yassir wrote: hi i have configured an asterisk server which i have tested locally with x-lite and that's ok but when i wanted to access to it since internet that hasent taken place knowing that my server has access to internet by a wifi router that has a public ip address (e-g a.b.d.c) and asterisk server has a private ip 192.168.1.111 http://192.168.1.111 (the firewall is disabled) can some one tellme how i should configure the x-lite clients with this configuration and what should i change to access to my server since internet -- Scales of success are not easy to be ridden ___ --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] Asterisk to Cisco 2600 GW DTMF Not Working
Sounds to me like inband vs rfc2833 issues. I found that one has to use the same codec throughout in order to make DTMF function and then use inband. This in turn forces you down the road of alaw or ulaw codecs. On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote: Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to transmit frame type 1024, while native formats is 4 (read/write = 4/4) Jun 26 17:53:52 WARNING[14248]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (160)? Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received The call continues with this error until I hang up. I have been adjusting the dial-peer dtmf settings in the 2600 and have tried all the dtmf settings in Asterisk. Any guidance will be appreciated. Thanks. JR ___ --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] Need to increase call count
It turns out that we were/are having trouble with out uplink On Tuesday 19 June 2007 05:39:27 am Dave Bour wrote: Have you tested the actual throughput on the link? What's it max out... What kind of latency are you seeing as it gets loaded. Can you do a local call to your own internal network (softphone or hardphone) as the system gets loaded and play the same file. Do you have quality issues. This will help isolate if it's a network or server issue. Shouldn't be a server issue at that load...but never hurts to look. D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Diehl Sent: Tuesday, June 19, 2007 12:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Need to increase call count Hi all. I've got a project where I need to make outbound calls and play a prerecorded .wav file to the called number. So far, I've only been able to make about 15 concurrent calls before the cound quality gets poor, and I really need to increase this. I've got QoS configured to prioritize IAX2 traffic above all and my connection to the Internet is a PtoP 100Mb ethernet link. (255.255.255.252 subnet mask) The server is an AMD Athlon(tm) 64 Processor 3400+ with 512Mb of RAM. The Nic isn't sharing an IRQ with anything else and the CPU never exceeds 15% utilization. Any ideas where I can look for improvement? TIA, -- Mike Diehl ___ --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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Diehl ___ --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] access to asterisk server since internet
See http://www.dyndns.com/services/dns/dyndns/ You can establish a name like yassir.dyndns.org that will point to the dynamic ip address for your Asterisk server. You should be able to use this for the domain for your VoIP service provider. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of skalli yassir Sent: Friday, June 22, 2007 11:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] access to asterisk server since internet hi i have configured an asterisk server which i have tested locally with x-lite and that's ok but when i wanted to access to it since internet that hasent taken place knowing that my server has access to internet by a wifi router that has a public ip address (e-g a.b.d.c) and asterisk server has a private ip 192.168.1.111 (the firewall is disabled) can some one tellme how i should configure the x-lite clients with this configuration and what should i change to access to my server since internet -- Scales of success are not easy to be ridden ___ --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] Nokia N95 + Dial Plan
I just had a similar problem and solved it with..
[from-internal-intldial]
exten = _+61X,1,Goto(from-internal,0${EXTEN:-9},1)
exten = _X.,1,Goto(from-internal,${EXTEN},1)
And put the E65 into the new contect. The first line stops all +61 (my
default country in australia)
The second catches everything else (like local 1300, 1800 etc numbers
that are in the phone without a +)
I can't find anywhere that tells me that the + is a valid pattern in a
dialplan but it works :-)
Obviously you could add any country codes you wish. I couldn't think of
something that would work for all codes as they are differing lengths.
Regards
Kevin
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PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
Please consider the environment before printing this e-mail
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Monday, 25 June 2007 1:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Nokia N95 + Dial Plan
Hello All,
Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XX for long distance
calls.
Is there anyway to create a + sign dial plan which will allow them to
dial a number with + sign.
Cheers,
Nitesh
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[asterisk-users] Module '***.so' did not register itself during load
Hi, I've experiencing this kind of problem. Actually, my asterisk is running perfectly. I've tested it, and I called some computer in my LAN. Then I enter the CLI and entered these commands - show modules - modules status (or so.. I forget) - restart now After I enter the last command, the CLI is exiting and nothing happened. Then I try to run the asterisk with command - asterisk But then nothing is happened. I check with ps command, and realize that asterisk isn't running at all. When I check the log, i got this kind of message : [Jun 27 08:58:17] WARNING[1255] loader.c: Module 'format_pcm_alaw.so' did not register itself during load [Jun 27 08:58:17] WARNING[1255] loader.c: Module 'chan_modem_aopen.so' did not register itself during load [Jun 27 08:58:17] WARNING[1255] loader.c: Module 'app_setcidname.so' did not register itself during load [Jun 27 08:58:17] WARNING[1255] loader.c: Error loading module 'app_txtcidname.so': /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: option_priority_jumping [Jun 27 08:58:18] WARNING[1255] loader.c: Module 'app_cut.so' did not register itself during load [Jun 27 08:58:18] WARNING[1255] loader.c: Module 'app_setcidnum.so' did not register itself during load [Jun 27 08:58:18] WARNING[1255] loader.c: Error loading module 'chan_h323.so': libh323_linux_x86_r.so.1.17.1: cannot open shared object file: No such file or directory [Jun 27 08:58:18] WARNING[1255] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 27 08:58:18] WARNING[1255] loader.c: Module 'chan_modem_i4l.so' did not register itself during load [Jun 27 08:58:18] NOTICE[1255] cdr_radius.c: Cannot load radiusclient-ng configuration file /etc/radiusclient-ng/radiusclient.conf. Is anybody know what happened? Because I really don't understand this. Please, any body, help me :( Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ ___ --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] Transfer Call to Cell Phone
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just dials out which ever line is not busy. Thanks!
I'm quite new to *, but I've got this in place in my first rendition, and
I'm pretty sure it does what you want:
exten = 101,1,Dial(SIP/${EXTEN},15,t)
exten = 101,n,Dial(Zap/4/12185551212,30,tpm)
exten = 101,n,VoiceMail([EMAIL PROTECTED])
exten = 101,n,Playback(vm-goodbye)
exten = 101,n,Hangup
caller dials extension 101. It first tries his desk for 15 seconds, then
it tries his cell over a zap channel (the 'p' turns on call screening),
then it finally hits voicemail. In our actual dialplan, the cell phone
call goes out over sip, so the line looks like this:
exten = 101,1,Dial(SIP/lesnet/12185551212,30,tpm)
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Hope that helps.
Ryan
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Re: [asterisk-users] Asterisk GUI
In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I have written a few jokes for this list over the years - it's nice to know that some people find them funny. PaulH ___ --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] kore dump
just an idea, but maybe qmail, samba, and bind have a smaller memory footprint than an in-use asterisk? No, probably not. Asterisk's is about 20-40 MB depending on the number of extensions, etc. Smbd's is similar, bind's is actually 90 MB (with about 600 zones). can you take the hardware offline long enough for a memtest? The machine has been retired (routine upgrade cycle). But I hardly doubt that was the problem. My guess is it was somehow related to limited CPU power (thread switching, interrupts, or whatnot). The old hardware was single CPU and a lot slower. --Luki ___ --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] Ring the second line when 1st line is busy
Isn't this what you are looking for?
http://voipspeak.net/index.php?option=com_contenttask=viewid=72Itemid=28
On 6/26/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Do any one any clue.
This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you
can see the extensions asssigned to a user as.
555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
8555 --- Line 2 -- Extensions which registers with SIP(Asterisk) -- User A
So when now someone calls one Extension 555 to User A, scenario as below.
1) If he is busy on the first line Ext 555, then ring Line 2 Ext 8555, if
no response send to voicemail
2) If he doesnt picks the line 1 Ext 555, then send to voicemail rather
then ringging on his second line ie Ext 8555.
This is what I need, if I can dow it with Follow me, then how, if through
ring group how.
*Deepak Naidu [EMAIL PROTECTED]* wrote:
Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone
on line 1(555) after 10 seconds then ring the line 2(8555). But this is
not a standard telephony which I have been advised to change like below.
If someone calls Ext 555 its busy(means on a phone with someone), then
only ring the second line ie 8555 even if that is busy send on voicemail.
If the first line 555 is free no one picks up then let it go on the
voicemail not second line, bcos now no one has picked the phone nor busy.
I could find anyway to do in FreePBX, so was wondering how about doing
this. Thanx for any input.
exten = 555,1,Macro(exten-vm,555,555)
exten = 555,n,Hangup
exten = 555,hint,SIP/555
exten = ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
--
Deepak
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