[Asterisk-Users] Starting Development Perl or Python

Hi guys, >From the drift of the mailing list most people seem to use perl for their AGI scripts. I personally have more experience with Python. Could you please advise why or if Perl is more suitable? Is it faster? Better supported? More documentation? Or any other comments. Our environmen

Re: [Asterisk-Users] best low-bandwidth strategy

Hi,   Given the following setup   Cisco (7960, G729) --> Asterisk --> IAX2(SPEEX)   It seems that asterisk cannot do the conversion (calls are rejected) even if i have G729 licenses   Is there a parameter in IAX.conf that allows this conversion.   PS if the call originates from FXS, i.e. FXS->Aster

[Asterisk-Users] best low-bandwidth strategy

Hi, To push voice through a long thin wan (dsl) there are two choices: (1) have the cisco's (7912G) talk g729a to each other (reinvite=yes), or (2) have the cisco's talk to their local * in ulaw (reinvite=no), which talk to each other through a more advanced low-bandwidth codec (ilbc or speex)

Re: [Asterisk-Users] SIP / GrandStream Configuration

On Thu, Sep 25, 2003 at 12:33:02AM -0400, Uriel Carrasquilla wrote: > Adam: > I believe you. I assume that the RTP is creating a symetric configuration > between * and the SIP phone. The situation we are left to live with is that > * (won't be the Sip phone) can only live in the Internet brave wo

[Asterisk-Users] Group pickup codes, etc.

Hi people, I've got Asterisk running nicely with two 7960s and hooked into our MD110 via a cisco AS5300. All is wonderful with the world...except, what is the deal with features like group pickup and so on... I have no idea what codes are available, what they do, etc...is there either a standar

Re: [Asterisk-Users] VIA vs Intel

Have you tried ALSA? I am just getting my platform up with the 1GHz C3. Seems to be running fine with 4 BT101s in a meetme conference using ULAW. Next is to try ALSA to get local console sound support. Yes, OSS_Lite does fail. AC97 driver does not recognize the chip ID. > I have. Heads up o

RE: [Asterisk-Users] SIP / GrandStream Configuration

Adam: I believe you. I assume that the RTP is creating a symetric configuration between * and the SIP phone. The situation we are left to live with is that * (won't be the Sip phone) can only live in the Internet brave world (and not behind a firewall). is this acceptable? Uriel -Original M

RE: [Asterisk-Users] SIP / GrandStream Configuration

On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: > Adam: > in reference to my first message, the NAT on the SIP/GS (a D-Link router) > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being > forwarded to the Sip/GS. > The Asterisk server, also behind another NAT (Linksys), has

Re: [Asterisk-Users] Purchasing Grandstream Phones

and IAX. Alas, i've offered to develop an IAX implementation for them, but all of my requests have gone unanswered. Jeremy McNamara Gary wrote: I would probably be interested except when will their products actually support GSM codecs ?? On Wed, 24 Sep 2003 21:34:36 -0500 (CDT), Dave Weis wrote

Re: [Asterisk-Users] VIA vs Intel

I have. Heads up on the built-in sound. Like everything else on the motherboard, it uses a VIA chipset and chan_oss will not work with it. Several posts have been made to the list in the past about the VIA chipset sound cards. Take a look at the Google archives for more info. Does anyone have

Re: [Asterisk-Users] Purchasing Grandstream Phones

Dream on!... as of now they dont. On Thu, 25 Sep 2003, Gary wrote: > > I would probably be interested except when will their products actually > support GSM codecs ?? > > On Wed, 24 Sep 2003 21:34:36 -0500 (CDT), Dave Weis wrote: > > > > >On Thu, 25 Sep 2003, Aaron Martin wrote: > >> Does anyone

Re: [Asterisk-Users] Purchasing Grandstream Phones

I would probably be interested except when will their products actually support GSM codecs ?? On Wed, 24 Sep 2003 21:34:36 -0500 (CDT), Dave Weis wrote: > >On Thu, 25 Sep 2003, Aaron Martin wrote: >> Does anyone know of any reliable supplier for Grandstream phones? >> Now, I am pretty sure that

Re: [Asterisk-Users] Purchasing Grandstream Phones

http://www.chagres.net/products/voip/phones.html bkw On Thu, 25 Sep 2003, Aaron Martin wrote: > Does anyone know of any reliable supplier for Grandstream phones? > > I tried dealing with David Li from Grandstream, but after emailing him an order in > August, and asking how he wanted payment, I

Re: [Asterisk-Users] Purchasing Grandstream Phones

On Thu, 25 Sep 2003, Aaron Martin wrote: > Does anyone know of any reliable supplier for Grandstream phones? > Now, I am pretty sure that I haven't done anything to offend these > people, and I am pretty sure that Grandstream are not against the idea > of selling product.. Has anyone had a Grands

[Asterisk-Users] Purchasing Grandstream Phones

Does anyone know of any reliable supplier for Grandstream phones?   I tried dealing with David Li from Grandstream, but after emailing him an order in August, and asking how he wanted payment, I never got a reply...   James Ho from DGTimes was happy to give me pricing, but when I sent him an

Re: [Asterisk-Users] echo for 15 seconds

- Original Message - From: Chad R. Graham >For the first 15 seconds of a call I get echo on the ata 186 side only. I >assume after that the echo canceller kicks in but is there any way to make it >happen faster? Same thing here - except we're using Cisco 7960 and 7940 IP phones. We're

RE: [Asterisk-Users] Check and restart script..

yes, it is not cron but a daemon. Iactually got the suggestion from this list. You can get all the glory details from: http://cr.yp.to/daemontools.html Dr. Bernstein tools. I have been using it with asterisk successfully for 4 months. Regards, Uriel -Original Message- From: [EMAIL PROTEC

[Asterisk-Users] Voicemail doesn't hangup

I'm running the a very recent CVS version of asterisk on an RH9 machine. My problem is that my x100p takes about 10 seconds to detect a hangup. After that it takes about 10 more seconds for the the zaptel device to release the line. Here's an example of my console report: == Parsing '/var/spo

RE: [Asterisk-Users] SIP / GrandStream Configuration

Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URie

[Asterisk-Users] Removal of anti-spam responder

As others have said, everything I post to asterisk users, I get this anti-spam HTML saying please authenicate. This email indicates it's on behalf of [EMAIL PROTECTED] Can we please remove this person from the list. thanks, Adam ___ Asterisk-Users m

[Asterisk-Users] (no subject)

  Dear All, I am going to deploy a VOIP network here in Canada with nodes all over town. This is for long distance services and hence would need a good reliable solution. I have looked into * and am very interested in it with all the value-added features as well as its capability to do H323

[Asterisk-Users] echo for 15 seconds

Hello,   I am running asterisk with two X100P cards using a cisco ata 186 "MGCP" for phone connections.   For the first 15 seconds of a call I get echo on the ata 186 side only.  I assume after that the echo canceller kicks in but is there any way to make it happen faster?   I have read som

[Asterisk-Users] help asterisk call waiting X100P -> MGCP ata 186

I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk -> ATA 186 MGCP --> analog phone. How do I answer the call waiting beep.. Thanks, I appreciate any help. ___ Aster

Re: [Asterisk-Users] SIP / GrandStream Configuration

How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible - Original Message - From: "Uriel Carrasquilla" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SI

Re: [Asterisk-Users] VIA vs Intel

James Golovich wrote: On Wed, 24 Sep 2003, Steven Critchfield wrote: As a phone platform, it may be overkill, but I bet it could drive a TDM400P card and be able to handle GSM compression. The question then again is if it is worth the cost for basically a 4 port asterisk based device like th

RE: [Asterisk-Users] SIP / GrandStream Configuration

Very valuable help. It is now working like a champ. This is a solution with SIP--NAT---Internet---Asterisk. No problems here. What I would like to do next is to move Asterisk behind a NAT as follows SIP---NAT---Internet---NAT---Asterisk do I need a STUN server? is there a chance this could work

Re: [Asterisk-Users] VIA vs Intel

> The only (serious) problem I have with it is that I'm unable to make the > cards use the IRQs I want. I always get USB using the same interrupt as > the X100P adapter, and the general mantra is that one should avoid > that. If you know of a way to reassign interrupts in a saner manner, I'd > appr

Re: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization

Do you have any dtmfmode=inband in you sip.conf? Regards, Gus - Original Message - From: "Scott Stingel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, September 24, 2003 5:13 PM Subject: RE: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization > Lot's of people o

RE: [Asterisk-Users] Dlink DG-104S (chan_mgcp) and configuration w/Asterisk

This is what I have in my mgcp.conf [dlink] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=dynamic context=international nat=yes ;dtmf=inband disallow=all allow=g711 allow=ulaw callerid = Andrew Joakimsen <321> line => aaln/1 callerid = Andrew Joakimsen <322> line =>

[Asterisk-Users] Dlink DG-104S (chan_mgcp) and configuration w/Asterisk

I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: [EMAIL PROTECTED]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP add

[Asterisk-Users] Chan_capi accountcode.. (repost)

Hi, All my inbound calls have a blank account code in the CDR.. Where or what is the correct way to set the "accountcode=" setting when using chan_capi channels? Do I do it in capi.conf or extensions.conf with a setvar? or some other way.. Thanks.. --

RE:# [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0

Thank you. Is there any documenantation available abount installing and configuring chan_capi? __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Us

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs

You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens.

[Asterisk-Users] Packet8 sans DTA310

Has anyone gotten Asterisk to work with Packet8 *without* the DTA310? --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) This message has been 'sanitized'

RE: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization

Lot's of people on here use Redhat 9.0 - don't worry! The 100% utilisation sounds wrong, assuming that asterisk is actually handling any calls though. Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] URL:www.evtmedia.com

Re: [Asterisk-Users] VIA vs Intel

> "Steven" == Steven Critchfield <[EMAIL PROTECTED]> writes: Steven> On Wed, 2003-09-24 at 13:13, Jon Pounder wrote: >> speaking of VIA - has anyone on the list looked at or used these ? >> http://www.mini-itx.com/store/default.asp?c=2¤cy=2 >> >> various collection of via based boards and

[Asterisk-Users] More on"Callprogress"

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively s

Re: [Asterisk-Users] Festival Problems

> > I have exactly the same symptoms with app_festival and I > suspect that > > send_waveform_to_channel routine in app_festival.c doesn't work > > correctly. > > > > Festival works correctly since it sends wave file to asterisk, which > > saves it in cache. If I strip app_festival header in that f

Re: [Asterisk-Users] VIA vs Intel

>> >> I am running Asterisk on one of these (T100P taking the single >> PCI slot.) (EPIA M1 Mini-ITX Motherboard) >> >> http://www.hushtechnologies.com/default.asp?pageID=2&Lang=ENG >> >> I bought it via http://mini-itx.com/ >> > > How many concurrent calls have you run on this MB?? > > What c

AW: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0

Hi Jim, i had the same probs, and it seems to be bug/feature of i4l. I can not find anything in the code that would bring these messages to the top of ttyI:-( Or is there somebody who knows it better ??? ;-) Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:

Re: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0

Hi Jim, get chan_capi from www.junghanns.net/asterisk/ install the capi drivers for your card from ftp.avm.de/cardware best regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-

Re: [Asterisk-Users] VIA vs Intel

> > I am running Asterisk on one of these (T100P taking the single > PCI slot.) (EPIA M1 Mini-ITX Motherboard) > > http://www.hushtechnologies.com/default.asp?pageID=2&Lang=ENG > > I bought it via http://mini-itx.com/ > How many concurrent calls have you run on this MB?? What codecs are y

[Asterisk-Users] No ring tone while dialing out with AVM PCI2.0

Hi, I use an AVM FRITZ PCI 2.0 to dial out but although it works OK and places the call there is no ring or busy tone. Has someone figured out this problem? Thanks __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://siteb

Re: [Asterisk-Users] Prebuilt Asterisk

Contact Sean Robertson at NETXUSA for prebuilt system. [EMAIL PROTECTED] 1-864-271-9868 1-800-292-0728 - Original Message - From: "Mike Hjorleifsson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, September 24, 2003 1:43 PM Subject: [Asterisk-Users] Prebuilt Asterisk

Re: [Asterisk-Users] VIA vs Intel

On Wed, 24 Sep 2003, Steven Critchfield wrote: > As a phone platform, it may be overkill, but I bet it could drive a > TDM400P card and be able to handle GSM compression. The question then > again is if it is worth the cost for basically a 4 port asterisk based > device like the ATA186? I have

Re: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization

i run some system with * & rh 9.0 be sure to have latest updates (install them after installing redhat, and before installing *) and check to have mpg123 installed (you must get it on the mpg123 website), since redhat has a mpg321 replacement that won't work with * matteo. Il mer, 2003-09-24 alle

RE: [Asterisk-Users] Undocumented variables in chan_sip.c

I haven't actually read the code involved, but my *guess* would be that setting "srvlookup" to "yes" means that the NAPTR/SRV lookup procedure described in RFC 3263 is used to turn SIP hostnames into an IP addresses. It's also possible that it means that Asterisk will use the older, deprecated proc

Re: [Asterisk-Users] VIA vs Intel

I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2&Lang=ENG I bought it via http://mini-itx.com/ > On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: >> Has anyone successfully run

Re: [Asterisk-Users] VIA vs Intel

On Wed, 2003-09-24 at 13:13, Jon Pounder wrote: > speaking of VIA - has anyone on the list looked at or used these ? > http://www.mini-itx.com/store/default.asp?c=2¤cy=2 > > various collection of via based boards and cases and other goodies that go > along with them. > > They are cheap enough th

Re: [Asterisk-Users] Cisco 2600 and ASTERISK

This is my configuration of my cisco router and still it does not want to work :( Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname asterisk ! aaa new-model aaa authentication login default local enable secre

[Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I track

Re: [Asterisk-Users] Prebuilt Asterisk

Mike Hjorleifsson wrote: Does anyone sell a preinstalled asterisk server ? I believe TelAppliant.co.uk will sell you a system called mypbx, which is basically just that. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asteris

Re: [Asterisk-Users] Prebuilt Asterisk

http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934130944.htm bkw On Wed, 24 Sep 2003, Mike Hjorleifsson wrote: > Does anyone sell a preinstalled asterisk server ? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.di

Re: [Asterisk-Users] VIA vs Intel

On Wed, 2003-09-24 at 10:41, Mike Hjorleifsson wrote: > Has anyone successfully run asterisk with a VIA processor ? > I have tried unsucessfully, do I have to run make with any specific switches > ? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTE

Re: [Asterisk-Users] VIA vs Intel

speaking of VIA - has anyone on the list looked at or used these ? http://www.mini-itx.com/store/default.asp?c=2¤cy=2 various collection of via based boards and cases and other goodies that go along with them. They are cheap enough they could work as either an asterisk server (diskless or with d

Re: [Asterisk-Users] VIA vs Intel

On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: > Has anyone successfully run asterisk with a VIA processor ? > I have tried unsucessfully, do I have to run make with any specific switches > ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible.

Re: [Asterisk-Users] Call transfert with dial plan

Do you think that my problem may be related to chan_vpb driver? Daniel Dunno.. I am not that close to the internal workings of Asterisk.. Sorry.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze _

[Asterisk-Users] Prebuilt Asterisk

Does anyone sell a preinstalled asterisk server ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] VIA vs Intel

Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml That covers the thridparty h323 stuff with * bkw On Wed, 24 Sep 2003, Sean Figgins wrote: > > That is about what I have been seing for help. Has anyone any clue what > to di with a 2600 that has a T1 adapt

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

This is inbound FXO's pointed at the autoattendat on our * server. On Wed, 24 Sep 2003, Doug Dimick wrote: > The configuration shouldn't be much different. Just replace port 1/0/0 > with port 1/0:23, or whatever the voice port your PRI/T1/E1 happens to be. > > PLAR is Private Line Auto Ringdown.

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

If you are using it with chan_h323 you need to set bearercap Speech on your voiceport. :P bkw On Wed, 24 Sep 2003, Sean Figgins wrote: > > That is about what I have been seing for help. Has anyone any clue what > to di with a 2600 that has a T1 adapter on a high-density high-density > voice por

[Asterisk-Users] Adding a DELAY to an ADSI script

I was searching through the app_adsi.c file and found some events and functions that are not used in the sample ADSI scripts. One of these functions is DELAY. I can't get this to work. Has anybody got this to work? I'm trying to create a HangUp soft key using the following code: KEY "Hangup" IS

Re: [Asterisk-Users] Check and restart script..

On Wed, 2003-09-24 at 11:57, WipeOut . wrote: > > Maybe you should look at init. From the init man page... > >When starting a new process, init first checks whether the file > >/etc/initscript exists. If it does, it uses this script to start the > >process. > > >

Re: [Asterisk-Users] Festival Problems

On Wed, 24 Sep 2003, Borut Senicar wrote: > I have exactly the same symptoms with app_festival and I suspect that > send_waveform_to_channel routine in app_festival.c doesn't work > correctly. > > Festival works correctly since it sends wave file to asterisk, which > saves it in cache. If I str

Re: [Asterisk-Users] Call transfert with dial plan

WipeOut . a écrit: So it is working sometimes and then sometimes it doesnt.. I haven't had this problem.. I am using the .81 firmware and I use the following transfer process.. I am using this firmware too 1. Press the transfer button. 2. Dial the extension that I want to transfer to. 3. Pres

Re: [Asterisk-Users] Festival Problems

I have exactly the same symptoms with app_festival and I suspect that send_waveform_to_channel routine in app_festival.c doesn't work correctly. Festival works correctly since it sends wave file to asterisk, which saves it in cache. If I strip app_festival header in that file I can play it. The pr

Re: [Asterisk-Users] Check and restart script..

> Maybe you should look at init. From the init man page... >When starting a new process, init first checks whether the file >/etc/initscript exists. If it does, it uses this script to start the >process. > >Each time a child terminates, init records the f

Re: [Asterisk-Users] Snom 200 errors?

WipeOut . schrieb: I have the same but everything still seems to be working so I haven't worried about it.. maybe there has been an extention to the SIP protocol?? ... me too. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.di

Re: [Asterisk-Users] Call transfert with dial plan

So it is working sometimes and then sometimes it doesnt.. I haven't had this problem.. I am using the .81 firmware and I use the following transfer process.. 1. Press the transfer button. 2. Dial the extension that I want to transfer to. 3. Press "Redial" button. (The "Redial" button has been ren

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

The configuration shouldn't be much different. Just replace port 1/0/0 with port 1/0:23, or whatever the voice port your PRI/T1/E1 happens to be. PLAR is Private Line Auto Ringdown. You pick up the port and it automatically dials. Think Batphone. In the configuration provided below it is configure

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: > This is simple to do.. > > voice-port

Re: [Asterisk-Users] Call transfert with dial plan

WipeOut . a écrit: What problem are you having with tranfer on the GS phone? I have no clearly defined situation but I have experienced some lost calls during a call transfert with GS. Regards, Daniel Hello, As I have problems getting transfert call working with my gran

Re: [Asterisk-Users] Check and restart script..

3 ways: 1. in /etc/inittab : d1:23:respawn:/usr/sbin/asterisk -fv 2. use daemontools from DJB (this is what I use) 3 safe_asterisk (maybe is better this way) :-) WipeOut . wrote: Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if

Re: [Asterisk-Users] SIP / GrandStream Configuration

have you tried to put nat=yes in the user definition in sip.conf ? Also, the * server is on a public IP? Matteo Il mer, 2003-09-24 alle 15:35, Uriel Carrasquilla ha scritto: > Hi there! > I installed the BudgetTone (GrandStream) on my LAN without any > problems. Then, I moved it to another loca

RE: [Asterisk-Users] Meridian Option 11 and asterisk

You can interface the two systems a variety of ways... the quick/easy route is a direct T1 from one switch to the other. You could do it via analog trunks too, but T1 signalling makes it so much smoother overall. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On B

RE: [Asterisk-Users] Festival Problems

Did you Answer the call before calling Festival? Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, September 24, 2003 11:13 To: [EMAIL PROTECTED] Run the command "festival" Give it the command (SayText "Would you l

Re: [Asterisk-Users] Check and restart script..

On Wed, 2003-09-24 at 10:01, WipeOut . wrote: > Has anyone written a script that can be used as a cron job or similar > that will test if Asterisk is running and if not restart it?? > > I have just had an issue where asterisk crashed and someone was trying > to call me.. it would be nice if it cou

RE: [Asterisk-Users] Check and restart script..

Something like this can tell if asterisk is running. You can modify it as needed. Doesn't match the "ps": if [ "A`ps -e | grep asterisk | grep -v grep`" = "A" ]; then echo echo "It's not running" echo else echo echo "It's running" echo fi Scott M. Stingel Emerging Voice Technology Inc. Email

RE: [Asterisk-Users] Check and restart script..

Could you not just add it to your /etc/inittab? -Original Message- From: WipeOut . [mailto:[EMAIL PROTECTED] Sent: 24 September 2003 16:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Check and restart script.. Has anyone written a script that can be used as a cron job or similar tha

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=b

RE: [Asterisk-Users] Check and restart script..

You can always use the "safe_asterisk" script...it's in the /usr/src directory. That's what I use. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 11:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users]

Re: [Asterisk-Users] Call transfert with dial plan

What problem are you having with tranfer on the GS phone? > Hello, > > As I have problems getting transfert call working with my grandstream > SIP Phones, I woul like to know if it is possible to do it with a proper > dial plan in exten.conf. > > I haven't found any information about that in t

Re: [Asterisk-Users] Festival Problems

Run the command "festival" Give it the command (SayText "Would you like to play a game?") Does it say anything? If not, then there's a problem Festival. Type (Quit) to quit the festival app. On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote: > I am trying to use festival (latest version 1.4.3) > I h

Re: [Asterisk-Users] THIS IS STRANGE

On Tue, 2003-09-23 at 09:44, Bartosz Jozwiak wrote: > Right now it works great! > Thanks so much. > > Could you tell me what is that: > 'canreinvite=no' in sip.conf ? > When SIP initiates the call, the INVITE message contains the information on where to send the media streams. * uses itself as

[Asterisk-Users] Check and restart script..

Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking

[Asterisk-Users] Call transfert with dial plan

Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTE

Re: [Asterisk-Users] Does SIP work?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 September 2003 16:36, Jamie Carl wrote: > Lack of documentation? > Welcome to the bleeding edge... I know, I just meant that pretty much everything else is either descriptive or described in sip.conf. Except the meaning of "[xxx]"

Re: [Asterisk-Users] Snom 200 errors?

I have the same but everything still seems to be working so I haven't worried about it.. maybe there has been an extention to the SIP protocol?? Later.. > The following error messages were observed in /var/log/asterisk/messages: > > Sep 22 10:26:42 NOTICE[1133735216]: File chan_sip.c, Line 5099

Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

Sorry about that: bt gives the following output: #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 #1 0x40150c61 in malloc () from /lib/i686/libc.so.6 #2 0x40157dd0 in strdup () from /lib/i686/libc.so.6 #3 0x0805603b in cfg_process (tmp=0x80ea890, _tmpc=0x47a6a26c, _last=0x47a6a270, buf=0x6

Re: [Asterisk-Users] Cisco 2600 and ASTERISK

Title: Message I want to make it work and document it. So if somebody could send me some information I will be very pleased.   Joe, what was your Cisco configuration ? - Original Message - From: Joseph Finley To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 20

RE: [Asterisk-Users] Does SIP work?

*This message was transferred with a trial version of CommuniGate(tm) Pro* Lack of documentation? Welcome to the bleeding edge... Enjoy.. J > -Original Message- > From: Tais M. Hansen [mailto:[EMAIL PROTECTED] > Sent: Wednesday, 24 September 2003 10:54 PM > To: [EMAIL PROTECTED

Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

I am running Mandrake 9.1 if that makes a difference. --- Patrick <[EMAIL PROTECTED]> wrote: > On Wed, 2003-09-24 at 15:41, jerk face wrote: > > Ok, here is the real gdb output. > > > > This GDB was configured as > > "i586-mandrake-linux-gnu"... > > Core was generated by `asterisk'. > > Program

RE: [Asterisk-Users] Cisco 2600 and ASTERISK

Title: Message I too would like to see it.  I've tried many times with the help of a few and never got it to work.  It always results in a fast busy.   Joe   -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz JozwiakSent: Wednesday,

Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

On Wed, 2003-09-24 at 08:41, jerk face wrote: > Ok, here is the real gdb output. > > This GDB was configured as > "i586-mandrake-linux-gnu"... > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > ... > ... > ... > > Loaded symbols for > /usr/lib/asterisk/

[Asterisk-Users] Snom 200 errors?

The following error messages were observed in /var/log/asterisk/messages: Sep 22 10:26:42 NOTICE[1133735216]: File chan_sip.c, Line 5099 (handle_request): Unknown SIP command 'PUBLISH' from '212.23.220.236' Sep 22 11:32:50 WARNING[1133735216]: File chan_sip.c, Line 4519 (handle_response ): Got 2

[Asterisk-Users] netconsole - bad file descriptor?

In looking through the /var/log/asterisk/messages log, I see about a million lines like: Sep 21 12:00:59 WARNING[1235176752]: File asterisk.c, Line 183 (netconsole): sel ect returned < 0: Bad file descriptor That was about the time I was attempting to test overhead paging using: ; the following p

Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)

On Wed, 2003-09-24 at 15:41, jerk face wrote: > Ok, here is the real gdb output. > > This GDB was configured as > "i586-mandrake-linux-gnu"... > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > ... > ... > ... > > Loaded symbols for > /usr/lib/asterisk/

[Asterisk-Users] X100P incoming calls - "hangup" delay

Hi All, I wasn't too sure how to word the subject, so I apologise for that. Anyway, I've got two X100P cards here accepting calls. Basically in Australia our ring cadence is 400ms on, 200ms off, 400ms on, 2000ms off, repeat. What I've noticed is that it takes about 8 seconds after the caller has

[Asterisk-Users] Cisco 2600 and ASTERISK

Hello,   Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it.   Regards,   -- bart

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