[asterisk-users] getting some segmentation faults with 1.8

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

At 08:13 PM 7/23/2010, you wrote: >This look to be a build problem with 1.8. We would need to see a copy >of your config.log and output from 'make install'. It is possible >your are loading old modules from 1.6 into 1.8. Check the timestamps >on these modules. So I went back to the 1.8 folder,

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

Just my 2 cents, Try this to see if it helps: - Try removing the Dahdi modules loaded into your kernel - Run /sbin/depmod - Reinsert the modules using modprobe [module name] - Restart Asterisk Ira wrote: > At 07:08 PM 7/23/2010, you wrote: > >> Rather then tell us it did not work, post a de

Re: [asterisk-users] Vocera Comm Badges

I've worked with these before. They are designed to run a whole hospital shift, so there should be no worries regarding the battery. I'm not aware of the server having any kind of SIP support -- I think you would need to have a PRI trunk to another PBX. The last time I talked to them they had th

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

On Fri, Jul 23, 2010 at 10:36 PM, Ira wrote: > WARNING[28505] loader.c: Error loading module 'chan_dahdi.so': > /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: > ast_smdi_interface_unref > WARNING[28505] loader.c: Error loading module 'app_stack.so': > /usr/lib/asterisk/modules/app_stac

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

> WARNING[28505] loader.c: Error loading module 'app_stack.so': > /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister This is the gosub issue. It's in app_stack. -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

At 07:08 PM 7/23/2010, you wrote: >Rather then tell us it did not work, post a debug log showing the issue. > >A side from that did you read the UPGRADE.txt and CHANGES file located >in the source directory? At least to see if anything seemed to mention gosub or DAHDI. Had there been some obvious

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

On Fri, Jul 23, 2010 at 8:18 PM, Ira wrote: > Is there something really basic I missed to get 1.8 to work? > Rather then tell us it did not work, post a debug log showing the issue. A side from that did you read the UPGRADE.txt and CHANGES file located in the source directory? -- Paul Belanger

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

On Fri, Jul 23, 2010 at 6:23 PM, bruce bruce wrote: > I am having this issue with PRI. But I do not use conference rooms. Our > system is a simple queue and extensions. > You will then need to enable PRI debugs and check the IE for disconnect. The see why Asterisk is not hanging up the channel.

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

On 7/23/10 6:18 PM, Ira wrote: > At 02:58 PM 7/23/2010, you wrote: >> The Asterisk Development Team has announced the release of Asterisk >> 1.8.0-beta1. > So being the brave type, I downloaded and installed this onto my > Asterisk Box. Compiled fine and installed fine, but it didn't work. > > I

Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

At 02:58 PM 7/23/2010, you wrote: >The Asterisk Development Team has announced the release of Asterisk >1.8.0-beta1. So being the brave type, I downloaded and installed this onto my Asterisk Box. Compiled fine and installed fine, but it didn't work. I kept getting errors on gosub and none of my

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

qpage -s snppserver.example.com -p lyle -f lyle test page AMARDEEP SINGH wrote: > Do you have working script? > > On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese > wrote: > > Maybe you need to read the man page for qpage. The qpage client can > send the page to

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Do you have working script? On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese wrote: > Maybe you need to read the man page for qpage. The qpage client can > send the page to an SNPP server over TCP/IP. > > Lyle > > AMARDEEP SINGH wrote: > > Our SMS-gateway is not PSTN accessible. > > > > On Thu, Jul

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

I am having this issue with PRI. But I do not use conference rooms. Our system is a simple queue and extensions. -Bruce On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet < mauf...@adinet.com.uy> wrote: > You're right but it do not detect that I hungs on my side of the line. > I think that

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

You're right but it do not detect that I hungs on my side of the line. I think that in some way we are going into a conference in some unwanted way with the two dadhi channels and when i hang up both lines stay bridged. I think that the trouble appears when i dial a number in an analog phone, hoo

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

Well, what about PRI? Why should this stay on? Isn't the native bridge just a bridge channel that should go down automatically if the actually Dahdi/ZAP channel is down and there are no SIP channels on either? Thanks, Bruce On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen wrote: > On Fri, Jul 23,

[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All intere

Re: [asterisk-users] voicemail

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias >Subject: [asterisk-users] voicemail * paraphrasing OP * Can I add these functions to voicemail? Some companies have the ability to press 6 to call back the people who have left a m

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote: >I guess same trouble with Elastix 1.5.2-2.3 >dahdi 2.1.0.4 19 > Asterisk 1.4.25.1 > Digium TDM 2400 That's an analog card. With an analog trunk, you're not guaranteed to know if the remote CO has hung up the lin

Re: [asterisk-users] Attended Transfer question

Hi! >> I've been asked to implement the following transfer workflow in an >> asterisk system, and I'm not seeing an easy way to do the bolded steps >> below (steps 4 and 5 for those with a text-only email client): > You could create a dynamic meetme room for the 3 legs and drop out when > done.

[asterisk-users] voicemail

Can i add functions to voicemail Like some companies have the ability to press 6 to call the people ho have leave a message Or When a people leave a message Press e.g 3 to mark the message -- _ -- Bandwidth and Colocation Prov

Re: [asterisk-users] Poor-man's paging through multiple phones?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Pauly Sent: Friday, July 23, 2010 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Poor-man's paging through m

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

I guess same trouble with Elastix 1.5.2-2.3 dahdi 2.1.0.4      19 Asterisk 1.4.25.1 Digium TDM 2400 [Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: master: 14, slave: 1, nothingok: 0 [Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Stopping tones on 14/0 talking to 1/0 [Jul 23 16:47:22] DEBUG[18890

Re: [asterisk-users] Attended Transfer question

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby >Subject: [asterisk-users] Attended Transfer question >I've been asked to implement the following transfer workflow in an asterisk system, and I'm not seeing an easy way to

[asterisk-users] Attended Transfer question

I've been asked to implement the following transfer workflow in an asterisk system, and I'm not seeing an easy way to do the bolded steps below (steps 4 and 5 for those with a text-only email client): 1 - Put the call on hold 2 - Call the extension for the staff member needed 3 - Give them a rundo

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Maybe you need to read the man page for qpage. The qpage client can send the page to an SNPP server over TCP/IP. Lyle AMARDEEP SINGH wrote: > Our SMS-gateway is not PSTN accessible. > > On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese > wrote: > > AMARDEEP SINGH wrot

Re: [asterisk-users] POE Splitters

Sounds like a great ear warmer!!! Hell, you can probably grill a panini with it if you're patient. On 7/23/10 6:39 AM, Matt wrote: You're using phones that draw 15Watts?!?! Let me know what brand this is so I can stay away from them. On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons mailto:d..

Re: [asterisk-users] 488 Not Acceptable Here

A packet capture would be most useful. Then, you could review your SDP with your provider. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Beak Sent: Friday, July 23, 2010 7:27 AM To: Asterisk Users Mail

[asterisk-users] Asterisk 1.4.34 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.4.34. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible wi

[asterisk-users] Asterisk 1.6.2.10 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.6.2.10. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possibl

Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

Our SMS-gateway is not PSTN accessible. On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese wrote: > AMARDEEP SINGH wrote: > > Hello All, > > Scenario: > -We use asterisk as voicemail server for our cellular network. Asterisk box > is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip. > -Extens

Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

Then I would suggest using the method I mentioned earlier, i.e. using macros. I have a really sophisticated dialplan for my multi-tenant system, which also incorporates some serious security stuff, along with call routing, trunk selection decisions and other checks, and for me macros work really we

Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

Le 23/07/2010 16:44, Zeeshan Zakaria a écrit : > Hi, > > I try to avoid any warnings, as they can turn into errors later. well, that's exactly the point of this inquiry :) > > I remember having problems with GoSub long time ago, don't remember > what it was, but I moved to macros after that. >

Re: [asterisk-users] POE Splitters

On Thu, 2010-07-22 at 17:41 -0500, Karl Fife wrote: > >> enough amps to power the full load at the end. > >> > > You could do someting with passive POE--in other words not 802.2af POE, but > rather the 'dumb' kind of POE which just injects power on the unused pairs. > Passive POE (being passive

Re: [asterisk-users] Dahdi dial plan

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Schwardt >Subject: [asterisk-users] Dahdi dial plan >can anybody please show me a valid dial plan for a dahdi card with a bri port? I can not get asterisk 1.6 to dial a number.

Re: [asterisk-users] Dahdi dial plan

On Fri, Jul 23, 2010 at 04:55:57PM +0200, Sebastian Schwardt wrote: > Hi, > > > > can anybody please show me a valid dial plan for a dahdi card with a bri > port? I can not get asterisk 1.6 to dial a number. I want to receive one > call on the first b channel and dial another number on the seco

Re: [asterisk-users] Poor-man's paging through multiple phones?

There is a 'Page' command in asterisk for this purpose. What you are trying to achieve, I have implemented a few times using MeetMe. But I needed to send a sip-info message to customers' grandstream phones to turn speakers on their speaker on. Do you have some similar option on your phones? Zeesh

Re: [asterisk-users] POE Splitters

On Fri, 23 Jul 2010, Matt wrote: > It's not necessarily this simple. There is an approximately 50-75foot cable > run through ceilings and walls (CAT5) to the location where the phones will > be. At the phone location there is no power. Why not use analogue phones? Get some nice ones with caller

[asterisk-users] Dahdi dial plan

Hi, can anybody please show me a valid dial plan for a dahdi card with a bri port? I can not get asterisk 1.6 to dial a number. I want to receive one call on the first b channel and dial another number on the second b channel of the same isdn port. I tried something like Dial(DAHDI/g1/12345

Re: [asterisk-users] POE Splitters

On 07/23/2010 04:40 PM, bruce bruce wrote: > You can also use Ethernet Over Power Lines solution or wireless :-) His issue wasn't getting the network connection delivered, it was the power :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsvill

Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

Hi, I try to avoid any warnings, as they can turn into errors later. I remember having problems with GoSub long time ago, don't remember what it was, but I moved to macros after that. For what you are trying to achieve, I use macros. Just jump to a macro, evaluate what you need to, save the res

Re: [asterisk-users] Poor-man's paging through multiple phones?

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >Subject: [asterisk-users] Poor-man's paging through multiple phones? >We have informacast, but it is too cumbersome for the users. >I know Asterisk can ring several phones at the same time... if one of >them answers, the othe

Re: [asterisk-users] POE Splitters

You can also use Ethernet Over Power Lines solution or wireless :-) On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg wrote: > On Fri, Jul 23, 2010 at 8:46 AM, Matt wrote: > > It's not necessarily this simple. There is an approximately 50-75foot > cable > > run through ceilings and walls (CAT5)

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

This is running Elastix (FreePBX), so I am pretty sure there is Hangup() requested. At least this doesn't happen ALL THE TIME. So, something is getting stuck. Thanks, Bruce On Fri, Jul 23, 2010 at 9:10 AM, Paul Belanger wrote: > On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce wrote: > > Any help

Re: [asterisk-users] ringback tone after MOH, before queue member bridged

I normally work with other 3rd party IVRs, usually once the Agent is Reserved we signal the phone system to play Music on Hold while it's ringing the Agent. The trick here is to replace the Music on Hold with a fake ring file. -Original Message- From: asterisk-users-boun...@lists.digium

Re: [asterisk-users] Vocera Comm Badges

Hi! > I´ve seen them at trade shows, I think I remember it being proprietary. > What about using Dect handsets? That Star Trek device has always interested me. Too bad they chose WiFi over DECT, though. Vocera badge: * WLAN b/g * Talktime 2-2.5 hours, standby 20-27 hours * headset jack * OLED d

[asterisk-users] Poor-man's paging through multiple phones?

We're mostly Cisco CallManager with some SIP and Asterisk. I want someone at one of our locations to be able to dial and number and have Asterisk simultaneously dial several Call-Manager extensions which are set to auto-answer and talk into the phone creating a sort of paging system. We have info

Re: [asterisk-users] [AsteriskNow] Errors withcleaninstall(onmainscreen when making calls)

No value was set in /etc/asterisk/cdr.conf added it, but without succes Long shot, but is loguniqueid=yes in your cdr.conf? > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "a

[asterisk-users] calls don't hang up correctly on VM

Hello List, I'm moving my asterisk testing installation from CENTOS 5.4 on a real machine to SUSE on a xen VM. Everything seemed to go off without a hitch until I really looked at it. The call answers and processes correctly, but when it is time to end the call, the phone never d

Re: [asterisk-users] [AsteriskNow] Errors withcleaninstall(onmainscreen when making calls)

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert Scholtalbers Sent: Friday, July 23, 2010 8:27 AM Long live diff. Yup it seem it would. Would there be a temp solution to route all errors to a file in /var/log/op_server or so In

Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

Long live diff. Yup it seem it would. Would there be a temp solution to route all errors to a file in /var/log/op_server or so In that why the screen stays clean until the source of the warnings/errors is found. We think it has to do with a setting in the config file, maybe all options have to be e

Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce wrote: > Any help is appreciated. > Are you explicitly calling Hangup() within your dial-plans? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com --

Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes

On 07/23/2010 11:17 AM, Alexander Aksarin wrote: > On 21:46 Thu 22 Jul , Steve Underwood wrote: > >> It might help if you explained what you expect those pages should look >> like. I see three quite plausible pages. >> > I expect to see this http://imagebin.ca/img/Eihpy0.jpg > > T

Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

Sorry my changes didn't solve the problem; for reference, this is what I did: 3360,3367c3360 < my $key; < if ( "$server^$hash_temporal{'Destination'}") { <$key = "$server^$hash_temporal{'Destination'}"; <} < else { <$key="0"; <

Re: [asterisk-users] POE Splitters

On Fri, Jul 23, 2010 at 8:46 AM, Matt wrote: > It's not necessarily this simple.  There is an approximately 50-75foot cable > run through ceilings and walls (CAT5) to the location where the phones will > be.  At the phone location there is no power. You always have options. You just have to decid

Re: [asterisk-users] POE Splitters

On 07/23/2010 02:46 PM, Matt wrote: > It's not necessarily this simple. There is an approximately 50-75foot > cable run through ceilings and walls (CAT5) to the location where the > phones will be. At the phone location there is no power. I thought it was fairly obvious, but search for "PoE extr

Re: [asterisk-users] POE Splitters

It's not necessarily this simple. There is an approximately 50-75foot cable run through ceilings and walls (CAT5) to the location where the phones will be. At the phone location there is no power. On Thu, Jul 22, 2010 at 11:33 PM, David Backeberg wrote: > On Thu, Jul 22, 2010 at 2:46 PM, Matt

Re: [asterisk-users] POE Splitters

You're using phones that draw 15Watts?!?! Let me know what brand this is so I can stay away from them. On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons wrote: > There is no such device -- it's outside of the POE spec. > > Class 3 devices are allowed to consume at max 15.4W. Most phones are class >

Re: [asterisk-users] Vocera Comm Badges

I've seen them at trade shows, I think I remember it being proprietary. What about using Dect handsets? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). _

Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

Hi, I've sent the file to Danny personally. He had made some adaptations to it and returned it. Unfortunately not 100% successful warnings didn't disapear. Hopefully a new release will be there soon with less cryptic warnings. Greetings, Albert The file in question is probably part of Flash Ope

[asterisk-users] 488 Not Acceptable Here

Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them: snipsnipsnip---

[asterisk-users] Vocera Comm Badges

Hi, has someone ever got their hands on the Comm Badges from Vocera ( http://www.vocera.com/ ) and knows if they use anything standard and could work with asterisk, or does someone know an alternative to their really small, light devices? Regards, Andreas

[asterisk-users] (no subject)

Hi, I have a problem with a SIP trunk between Asterisk and central OXE Alcatel, especially sometimes are not received inbound calls with following messages: -- Executing [...@test:1] AGI("SIP/800-084250f8", "agi://127.0.0.1/test.agi") in new stack -- AGI Script agi://127.0.0.1/te

Re: [asterisk-users] Channels not coming up

On Fri, Jul 23, 2010 at 10:32:28AM +0100, Deepika Nijhawan wrote: > Hi, > > > > I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi > status is not showing alarms but channels are not coming up. It is not > showing any channels when i run 'dahdi show channels'. Could any

Re: [asterisk-users] Channels not coming up

On Fri, Jul 23, 2010 at 10:32:28AM +0100, Deepika Nijhawan wrote: > Hi, > > > > I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. dahdi_genconf only generates configuration. It does not apply it. Is there any demand for a script like genzaptelconf that also applies it? I

Re: [asterisk-users] Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?

Thanks. it is depends on mysqlclient.so. after i installed this module. it's ok. 2010/7/22 Gareth Blades : > Zhang Shukun wrote: >> hi,list >>       Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?  after >> i make and make install. i cant find the .so file. >> >> is this mean it can't ins

[asterisk-users] Channels not coming up

Hi, I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi status is not showing alarms but channels are not coming up. It is not showing any channels when i run 'dahdi show channels'. Could anyone help pls. Thanks Deepika -- __

[asterisk-users] ringback tone after MOH, before queue member bridged

Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queu

Re: [asterisk-users] MOH distorted voice in Native and MP3 format

> Something you may want to try (its fixed it for us) is putting an I > (uppercase I) on the asterisk invocation line. > > We run servers in the cloud and can't get reliable timing from ISDN > cards etc so this instructs asterisk to generate its own internal > timing. If you have ISDN you probably