Re: [Asterisk-Users] How to set busy
use groups, check the commands/functions group and checkgroup. On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available lines, but if the call was routed via [singletrackmind] dial would return busy if the channel already had one call. Thanks Miles ___ --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
Re: [Asterisk-Users] How to set busy
On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. I suppose incominglinit=1 in the sip.conf of that phone works exactly the wrong way round? -Benoit- ___ --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
[Asterisk-Users] oh323.conf problem
I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I have tested several codec's with several frames, but I haven't find combination that works. I'm using Asterisk-1.2.6, pwlib_Mimas_patch2, openh323_Mimas_patch2 and asterisk-oh323-0.7.3 channel driver. Below is the conf file and CLI output. -- Tomislav Parcina *1 Oh323.conf [general] listenAddress=0.0.0.0 listenPort=1720 outboundMax=100 inboundMax=100 simultaneousMax=100 context=sip [register] alias=asterisk alias=123 [codecs] ; no codec defined *2 == Parsing '/etc/asterisk/oh323.conf': Found Apr 9 10:37:57 WARNING[4015]: chan_oh323.c:5008 reload_config: Category [codecs ] not present in configuration file (oh323.conf). Apr 9 10:37:57 NOTICE[4015]: chan_oh323.c:5227 load_module: No codecs configure d! Disabling H.323 channel driver. *3 [general] listenAddress=0.0.0.0 listenPort=1720 outboundMax=100 inboundMax=100 simultaneousMax=100 context=sip [register] alias=asterisk alias=123 [codecs] codec=G711A frames=20 *4 [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Illegal instruction [EMAIL PROTECTED] ~]# ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set busy
Benoit Panizzon wrote: On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. I suppose incominglinit=1 in the sip.conf of that phone works exactly the wrong way round? That will return CHANUNAVAIL instead of the needed BUSY for DAILSTATUS Thanks though. ___ --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
Re: [Asterisk-Users] How to set busy
C F wrote: use groups, check the commands/functions group and checkgroup. I guess I can see how this would be useful, but is there no way to get it to return BUSY in DIALSTATUS var? On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available lines, but if the call was routed via [singletrackmind] dial would return busy if the channel already had one call. Thanks Miles ___ --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
Re: [Asterisk-Users] How to set busy
Why not use the busy command, in combination with the groupcheck commands - refer to http://www.voip-info.org/wiki/index.php? page=Asterisk+cmd+Busy On 09/04/2006, at 5:01 PM, Miles Scruggs wrote: C F wrote: use groups, check the commands/functions group and checkgroup. I guess I can see how this would be useful, but is there no way to get it to return BUSY in DIALSTATUS var? On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available lines, but if the call was routed via [singletrackmind] dial would return busy if the channel already had one call. Thanks Miles ___ --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
[Asterisk-Users] Re: [asterisk-dev] Announcing Astmanproxy 1.20
On 04/08/06 11:26 [EMAIL PROTECTED] said the following: I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy we've just started using astmanproxy, and i'll soon be submitting a couple of patches which addresses the following: 1. Building astmanproxy on FreeBSD 2. having astmanproxy reconnect if asterisk dies and restarts who should i submit patches to ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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
[Asterisk-Users] how to communicate two PCs on LAN with Asterisk
Dear Asterisk users, I m working on a final year research based project on Asterisk ... the work I would like to take from Asterisk is to have voice conversation between two PCs connected with eachother on a LAN with no Internet connection by using minimum hardware ... plz if anyone can guide me how to doit with what hardware and software ...your help in thisconcern will be highly appreiciated .. Thankyou for your consideration Regards, Shahzad Khoja ___ --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
[Asterisk-Users] Disable 407 proxy authentication for outbound domains
Hello, I posted a lot of mails may be asterisk is not able to accept sip calls from internet !? My english is not fluent i try my best ! My problem I use ser+asterisk. For local calls there are no problem (PSTN or IP) Now i wish to receive calls from other internet domain but asterisk ask for authentication 407. IS IT possible to Disable authentication for incoming calls to my sip uri ? Look at my sip.conf and extensions.conf [general] context=sip bindport=5050 realm=nxs.yi.org bindaddr=nxs.yi.org [sip] include = info include = support exten = info,1,Answer() exten = info,2,Dial(Sip/84,10) exten = info,3,Dial(Sip/85,10) exten = info,4,Hangup exten = support,1,Answer() exten = support,2,Queue(support|t||) exten = support,3,Hangup Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com extensions.conf Description: 3949034846-extensions.conf sip.conf Description: 3455877249-sip.conf ___ --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
[Asterisk-Users] Re: [asterisk-dev] Announcing Astmanproxy 1.20
On 04/08/06 11:26 [EMAIL PROTECTED] said the following: I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy we've just started using astmanproxy, and i'll soon be submitting a couple of patches which addresses the following: 1. Building astmanproxy on FreeBSD 2. having astmanproxy reconnect if asterisk dies and restarts who should i submit patches to ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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
[Asterisk-Users] Re: [asterisk-dev] Disable 407 proxy authentication for outbound domains
On 11:08, Sun 09 Apr 06, [EMAIL PROTECTED] wrote: Hello, I posted a lot of mails may be asterisk is not able to accept sip calls from internet !? My english is not fluent i try my best ! My problem I use ser+asterisk. For local calls there are no problem (PSTN or IP) Now i wish to receive calls from other internet domain but asterisk ask for authentication 407. IS IT possible to Disable authentication for incoming calls to my sip uri ? Look at my sip.conf and extensions.conf [general] context=sip bindport=5050 realm=nxs.yi.org bindaddr=nxs.yi.org add this here: allowguest=yes and add a user section like: [guest] type=user insecure=very context=sip . -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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
Re: [Asterisk-Users] Problems with registering iaxy
On 9 Apr 2006, at 06:04, Bartosz Wegrzyn - asterisk wrote: Anyone knows hot to fix that? Thanks I used to have my iaxy registered to my old version of asterisk. I switched to 1.2 ver and now registration fails. my config for iax.conf for that client looks like this: [user] username=user type=friend context=sip auth=plaintext secret=password host=dynamic disallow=all allow=ulaw trunk=no I provisioned my iax with this config: [EMAIL PROTECTED] iaxyprov]# cat iaxy ; ; IAXY Provisioning description ; ;dhcp ip: 192.168.1.249 netmask: 255.255.255.0 gateway: 192.168.1.251 codec: ulaw ;codec: adpcm server: 192.168.1.251 ;altserver: 192.168.0.2 user: user pass: password When I do iax2 debug I see this: IAX2 Debugging Enabled Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 2ms SCall: 12640 DCall: 0 [192.168.1.249:4569] USERNAME: user REFRESH : 60 DEVICE TYPE : iaxy2 SERVICE IDENT : 0003640005a8 PROVISIONG VER : 3503263220 voip*CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00012ms SCall: 00011 DCall: 12640 [192.168.1.249:4569] AUTHMETHODS : 1 USERNAME: user voip*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 12640 DCall: 00011 [192.168.1.249:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 2ms SCall: 08797 DCall: 0 [192.168.1.249:4569] USERNAME: user REFRESH : 60 DEVICE TYPE : iaxy2 SERVICE IDENT : 0003640005a8 PROVISIONG VER : 3503263220 Any ideas what is wrong? Does new asterisk differs in the iax2 registration? Thanks Bart I don't have an IAXy, so I can't be sure, but it looks like your's is rejecting asterisk's REGAUTH message. The REGAUTH has AUTHMETHODS : 1 USERNAME: user So I'm guessing that it doesn't like the AUTHMETHOD - plaintext. try setting it to MD5 in iax.conf, it is good policy anyhow not to send plaintext passwords. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set busy
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
For multiline phones how do you set SIP channels to busy. For instance
if SIP/101 is on a call then dial would return busy. Right now it just
starts ringing on line X, and stacks up from there.
${DIALSTATUS} BUSY comes from the phone.
You can limit the possible lines to an phone with call_limit.
If you have call_limit=1 you will never get an BUSY from the phone.
If call_limit smaller or equal the maximum call to an phone, you will never
get an BUSY.
So use groupcount and make your own logic. Then you know whats going on.
best regards
Thomas
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Re: [Asterisk-Users] How to set busy
On Sunday 09 April 2006 08:46, Benoit Panizzon wrote: On Sunday 09 April 2006 06:02, Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. I suppose incominglinit=1 in the sip.conf of that phone works exactly the wrong way round? incominglimit and outgoinglimit is replaced by call_limit in Asterisk 1.2. ___ --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
Re: [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi, Sorry for my delay writting here. My SIP.conf is similar of yours, i only don't use qualify=yes, is it compulsory? I have 100 users and if i activate qualify it will increase the traffic in my network no? Best regards, Marco Mouta On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, If your sip.conf is not setup properly SJPhone will not work. Here is my SJPHpne SIP config: sip.conf** ... ;SJphone [410] context=longdistance ;canreinvite=no type=friend username=410 secret=passwd410 callerid=410 qualify=yes nat=no host=dynamic [EMAIL PROTECTED] disallow=all ;allow=g729 allow=gsm allow=ilbc allow=ulaw allow=alaw dtmfmode=rfc2833 Callgroup=1 pickupgroup=1 ... * Thanks Marco Mouta wrote: Windows XP service Pack 2 What you mean with SIP config look like? I've everything by default, only config for Calls through SIP proxy Bug patches from sjphone? On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, What does your SIP config look like for the SJPhone? Also what operating system does this PC have and is it up to date with security and bug patches. Thanks Marco Mouta wrote: Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta ___ --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 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10
On Saturday 08 April 2006 20:18, Colin MacMillan wrote: Hello, 6) From here I enter the qozap directory. cd qozap 7) now I get the following error - linux:/usr/src/bristuff-0.2.0-RC8q/qozap # insmod qozap.ko insmod: error inserting 'qozap.ko': -1 Invalid module format Any help is greatly appreciated. I'm no expert so sorry if this posting is too 'noob' for some. What happens when you type the following ? modprobe --show-depends qozap modprobe -v qozap Paul ___ --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
[Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles ___ --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
Re: [Asterisk-Users] how to communicate two PCs on LAN with Asterisk
KhojaS wrote: Dear Asterisk users, I m working on a final year research based project on Asterisk ... the work I would like to take from Asterisk is to have voice conversation between two PCs connected with eachother on a LAN with no Internet connection by using minimum hardware ... plz if anyone can guide me how to do it with what hardware and software ... your help in this concern will be highly appreiciated .. Thankyou for your consideration Regards, Shahza Get three PCs and a download and install asterisk on one of them. On the other two PCs install the OS of your choice and find a freeware softphone that supports SIP or IAX2 and install that. Configure the phones to point to the asterisk machine and then configure the asterisk machine to handle the phones. Use [EMAIL PROTECTED] and you should be able to take the next 364 days off. Thanks, Steve ___ --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
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? ___ --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
[Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles ___ --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
Re: [Asterisk-Users] meetme
Steve Totaro wrote: Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ --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
RE: [Asterisk-Users] Best ATA for general residential deployment??
Grandstreams are totally useless, I had to switch all my phones to Linksys. Grandstream will not even support you and their router side do not work for the 486 or 496. -- Original message -- From: "Andre Rodrigues (Cheyenne)" [EMAIL PROTECTED] I have more than 20 ATA 386. They can not work for more than one day without a local and "hard reboot". Do no buy these ata please!!! Regards Amr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: quarta-feira, 22 de Fevereiro de 2006 23:11 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Best ATA for general residential deployment?? Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -Mike Mi chael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED]-Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt <[EMAIL PROTECTED]>wrote:Yes.. there are provisioning tools that you have to get.Unfortunately it's this catch 22 loop. You have to prove that youcan offer 200+ ATAs to customers, or you can't get the tools, butyet, you don't really want to offer tho se ATAs to the customer'swithout having the tools. This sounds like yet another reason to avoid purchasing Sipuraequipment and supporting Sipura in any way. I don't know about youguys, but I have better things to do than screw around with asininevendor policies that make it more difficult than necessary to getthings done. True, but it's kind of a "pick your poison" situation in my opinion. Ht-486 anyone? ___ --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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: Steve Totaro wrote: Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? ___ --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
Re: [Asterisk-Users] How to set busy
Many multi-line phones allow you to use the same username/password for all lines. Then the phone only actually registers once using that username and password, not once for each line. What we do with the Polycoms is configure each line to register as a different username/password (we use the MAC address followed by a - and a letter to indicate which line.) As long as you turn off Call Waiting on the phone you can avoid all the checkgroup/setgroup and incominglimit options other people are talking about. Miles Scruggs wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available lines, but if the call was routed via [singletrackmind] dial would return busy if the channel already had one call. ___ --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
[Asterisk-Users] GXP-2000 and Voicemail
Hi, I have a few GXP-2000 working fine with Asterisk. The one thing I have not been able to do is to program the MSG button to dial the Voicemail extension. How can I program that button? I normally use extension for voicemail. Can anyone shed any light? Thanks, Waldo ___ --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
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches the first peer (for my provider) and
never matches the next two that I also want to use.
Seems that it will only do a match based on IP Address/Host and not
on accountname or incoming phone number.
The help I have recieved here has not really addressed the origial question
of how to get the calls to come directly into a context from the sip peer
itself, however they have pointed out some work-arounds to what asterisk
seemingly does not support doing directly.
If I am wrong with this conclusion please help me out!
I have been able to accomplish what is needed by simply having an
initial context that everything comes into (possible security issue)
and then immediately issue a Goto() to get the call into the context where
it belongs.
This 'feels' very hokey and wrong, but it works for now!
Thanks for the help!
Take care!
Steve
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1);
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten = 5185551234,2,Goto(custom-callid,s,1) ;
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the
sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do or explain the simple concept that I am
trying to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension
needs to
; be defined in extensions.conf to be able to accept calls from this
SIP
proxy
I actually need to do 3 of these.
;register = 2345:[EMAIL PROTECTED]/1234
;
;Register 2345 at sip provider 'sip_proxy'. Calls from this
provider
;connect to local extension 1234 in extensions.conf, default
context,
;unless you configure a [sip_proxy] section below, and configure
a
;context.
Ok I have 3 accounts from the same provider
one [sip_proxy] section just puts me in the same problem boat I'm
already
in using a register line
the calls some into the context specified in [general] section of
sip.conf
I need to somehow differentiate the three SIP 'lines' and give
them different contexts to start in.
;Tip 1: Avoid assigning hostname to a sip.conf section like
[provider.com]
OK sure then how will this associate with my register line that
uses provider.com
This makes no sense to me...
I mean It really makes no sense...
Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts
Hi,
If you don't specify a host= statement in sip.conf and you have a
section that includes a username and secret plus type=peer, it will
match on username and secret. (That implies that if you have three
different numbers registered with your sip provider all under one
username, calls for all three will match the first section in
sip.conf
that contains that username and secret.)
Thank you for this tidbit as well.
It seems that I need the host= to actually be there for it to work though.
I've always used the same [peer] for incoming and outgoing calls,
If I get rid of the host= outgoing calls of course stop working.
This seems to be a strong hint that I need to explore using seperate peers
for incoming and outbound calls.
Put all the incoming peers first so they are not matched by host first and
then have the others at the bottom for outbound calls.
I will give this a try,
Thanks!
Steve
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches the first peer (for my provider) and
never matches the next two that I also want to use.
Seems that it will only do a match based on IP Address/Host and not on
accountname or incoming phone number.
The help I have recieved here has not really addressed the origial question
of how to get the calls to come directly into a context from the sip
peer itself, however they have pointed out some work-arounds to what
asterisk seemingly does not support doing directly.
If I am wrong with this conclusion please help me out!
I have been able to accomplish what is needed by simply having an
initial context that everything comes into (possible security issue) and
then immediately issue a Goto() to get the call into the context where
it belongs.
This 'feels' very hokey and wrong, but it works for now!
Thanks for the help!
Take care!
Steve
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234);
exten = 5185551234,2,Goto(custom-callid,s,1);
On 4/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure that you have created context1 context2 and context3 in your
extensions.conf
And in this context1 context2 and context3 you must have handler for
1234; 2345; and 3456;
example:
[context1]
exten = 1234,1,Answer()
exten = 1234,2,Playback(vm-goodbye)
exten = 1234,3,Hangup()
I didn't test this code, but this is my tip the main idea is that you
need to catch de DID and make a GoTo for the context you want.
Best regards,
Marco Mouta
On 4/2/06, Rich Adamson [EMAIL PROTECTED] wrote:
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the
sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in
/usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them. They
do not detail, do or explain the simple concept that I am trying
to accomplish.
If they do I don't see it.
#1 I have more than one incoming SIP account
#2 I would like to have them come into the context of
my choice when a call comes in.
HOW do I do this?
currently I have 3 register lines
there is no way to specify in a register line
some way of making the call start in any other context
other than what is specified in the [general] section
of sip.conf
It seems that somehow maybe if there is a peer tat is somehow
matched to the register line (how???) it may work.
There may be some crazy way to do this within a peer
if so this is the information I am looking for...
The examples and descriptions are not at all clear to me
I have 3 accounts with the same provider
How do I get incoming calls to come into three different contexts
that I will create is the question.
From the example file I see:
Asterisk can register as a SIP user agent to a SIP proxy
(provider)
; Format for the register statement is:
; register =
user[:secret[:[EMAIL PROTECTED]:port][/extension]
;
; If no extension is given, the 's' extension is used. The
extension
needs to
; be defined in extensions.conf to be able to accept calls from
this
SIP
proxy
I actually need
RE: [Asterisk-Users] meetme
Snip.. Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? Also post a 'show applications' form your asterisk CLI prompt. ___ --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
Re: [Asterisk-Users] oh323.conf problem
I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I have tested several codec's with several frames, but I haven't find combination that works. I'm using Asterisk-1.2.6, pwlib_Mimas_patch2, openh323_Mimas_patch2 and asterisk-oh323-0.7.3 channel driver. Below is the conf file and CLI output. -- Tomislav Parcina *1 Oh323.conf [general] listenAddress=0.0.0.0 listenPort=1720 outboundMax=100 inboundMax=100 simultaneousMax=100 context=sip [register] alias=asterisk alias=123 [codecs] ; no codec defined *2 == Parsing '/etc/asterisk/oh323.conf': Found Apr 9 10:37:57 WARNING[4015]: chan_oh323.c:5008 reload_config: Category [codecs ] not present in configuration file (oh323.conf). Apr 9 10:37:57 NOTICE[4015]: chan_oh323.c:5227 load_module: No codecs configure d! Disabling H.323 channel driver. *3 [general] listenAddress=0.0.0.0 listenPort=1720 outboundMax=100 inboundMax=100 simultaneousMax=100 context=sip [register] alias=asterisk alias=123 [codecs] codec=G711A frames=20 *4 [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Illegal instruction [EMAIL PROTECTED] ~]# Hi, I have had the exact same problem last week. I have not yet solved it. So instead I am using ooh323, but would prefer to use oh323. Can anyone help? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
RE: [Asterisk-Users] Best ATA for general residential deployment??
Not true. There are hundreds of thousands of Grandstream adapters in use around the world. Grandstream support is not perfect, but it is as good or better better than most vendors, including Linksys/Sipura.The Grandstreams do currently have a bug with header compression right now that causes problems for some PPPoE setups, but it's getting fixed. The newest firmware is very stable overall. We work with almost every device and they all have some issues. If you consider the vast variety of different equipment that these things have to be interoperable with you can begin to appreciate how challenging it is to make them work properly. Given the dynamic nature of the environment, there will always be a certain number of situations where a given product doesn't perform. I stand by my original assertion. The Linksys line of products are also excellent, but they are considerably more expensive for the same functionality. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, April 09, 2006 9:12 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Best ATA for general residential deployment?? Grandstreams are totally useless, I had to switch all my phones to Linksys. Grandstream will not even support you and their router side do not work for the 486 or 496. -- Original message -- From: "Andre Rodrigues (Cheyenne)" [EMAIL PROTECTED] I have more than 20 ATA 386. They can not work for more than one day without a local and "hard reboot". Do no buy these ata please!!! Regards Amr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: quarta-feira, 22 de Fevereiro de 2006 23:11 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Best ATA for general residential deployment?? Absolutely. HT-486 is my pick for best all-around unit based on ease-of-use, value, performance and reliability. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best ATA for general residential deployment?? On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote: On 2/22/06, Matt <[EMAIL PROTECTED]>wrote: Yes.. there are provisioning tools that you have to get. Unfortunately it's this catch 22 loop. You have to prove that youcan offer 200+ ATAs to customers, or you can't get the tools, butyet, you don't really want to offer those ATAs to the customer'swithout having the tools. This sounds like yet another reason to avoid purchasing Sipuraequipment and supporting Sipura in any way. I don't know about you guys, but I have better things to do than screw around with asinine vendor policies that make it more difficult than necessary to get things done. True, but it's kind of a "pick your poison" situation in my opinion. Ht-486 anyone? ___ --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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ --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
Re: [Asterisk-Users] meetme
On Sun, 09 Apr 2006 09:12:42 -0700 Miles Scruggs [EMAIL PROTECTED] spake: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ Did you have the ztdummy and stuff compiled into the kernel before you compiled asterisk? If not, asterisk skips compiling the meetme application. ___ --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
Re: [Asterisk-Users] GXP-2000 and Voicemail
Look at the Account Settings for Voice Mail UserID. Hi, I have a few GXP-2000 working fine with Asterisk. The one thing I have not been able to do is to program the MSG button to dial the Voicemail extension. How can I program that button? I normally use extension for voicemail. Can anyone shed any light? Thanks, Waldo ___ --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
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ Is MeetMe listed if you type show applications in the CLI? Try specifying a room number in extensions.conf. Does it work without the dynamic stuff? MeetMe([confno][,[options][,pin]]) ___ --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
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ From the wiki: If you compiled Asterisk yourself from source and neither /usr/include/linux/zaptel.h nor /usr/local/include/zaptel.h were present (this will be the case if you add the Zaptel drivers after the fact.), certain applications (fzapras, meetme, flash, zapbarge, zapscan and page) will not have been compiled. ___ --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
[Asterisk-Users] txfax tiff file format
Dear folks, I got a problem sending faxes using spandsp. Primerily, when the tiff file made using GIMP 2 with different compresions the fax app break downs whole *. Moreover when i made a tiff file using Microsoft mdi, everything works fine but on the other end of the call, the received fax is shrinked in size. Anyone has any idea whats the right file format and compression type for it? PS. Im using libtiff-3.7.1-2 and spandsp-0.0.2-pre25 Regards. --- M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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
Re: [Asterisk-Users] meetme
Snip.. Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? Also post a 'show applications' form your asterisk CLI prompt. Here they are all 154 of them (and meetme is missing from the list) *CLI show applications -= Registered Asterisk Applications =- AbsoluteTimeout: Set absolute maximum time of call AddQueueMember: Dynamically adds queue members ADSIProg: Load Asterisk ADSI Scripts into phone AgentCallbackLogin: Call agent callback login AgentLogin: Call agent login AgentMonitorOutgoing: Record agent's outgoing call AGI: Executes an AGI compliant application AlarmReceiver: Provide support for receving alarm reports from a burglar or fire alarm panel Answer: Answer a channel if ringing AppendCDRUserField: Append to the CDR user field Authenticate: Authenticate a user BackGround: Play a file while awaiting extension BackgroundDetect: Background a file with talk detect Busy: Indicate the Busy condition ChangeMonitor: Change monitoring filename of a channel ChanIsAvail: Check channel availability ChanSpy: Listen to the audio of an active channel CheckGroup: Check the channel count of a group against a limit Congestion: Indicate the Congestion condition ControlPlayback: Play a file with fast forward and rewind Cut: Splits a variable's contents using the specified delimiter DateTime: Says a specified time in a custom format DBdel: Delete a key from the database DBdeltree: Delete a family or keytree from the database DBget: Retrieve a value from the database DBput: Store a value in the database DeadAGI: Executes AGI on a hungup channel Dial: Place a call and connect to the current channel Dictate: Virtual Dictation Machine DigitTimeout: Set maximum timeout between digits Directory: Provide directory of voicemail extensions DISA: DISA (Direct Inward System Access) DumpChan: Dump Info About The Calling Channel DUNDiLookup: Look up a number with DUNDi EAGI: Executes an EAGI compliant application Echo: Echo audio read back to the user EndWhile: End A While Loop EnumLookup: Lookup number in ENUM Eval: Evaluates a string Exec: Executes internal application ExecIf: Conditional exec ExecIfTime: Conditional application execution based on the current time ExternalIVR: Interfaces with an external IVR application Festival: Say text to the user ForkCDR: Forks the Call Data Record GetCPEID: Get ADSI CPE ID GetGroupCount: Get the channel count of a group GetGroupMatchCount: Get the channel count of all groups that match a pattern Gosub: Jump to label, saving return address GosubIf: Jump to label, saving return address Goto: Jump to a particular priority, extension, or context GotoIf: Conditional goto GotoIfTime: Conditional Goto based on the current time Hangup: Hang up the calling channel HasNewVoicemail: Conditionally branches to priority + 101 with the right options set HasVoicemail: Conditionally branches to priority + 101 with the right options set IAX2Provision: Provision a calling IAXy with a given template ICES: Encode and stream using 'ices' ImportVar: Import a variable from a channel into a new variable LookupBlacklist: Look up Caller*ID name/number from blacklist database LookupCIDName: Look up CallerID Name from local database Macro: Macro Implementation MacroExit: Exit From Macro MacroIf: Conditional Macro Implementation MailboxExists: Check to see if Voicemail mailbox exists Math: Performs Mathematical Functions MD5: Calculate MD5 checksum MD5Check: Check MD5 checksum Milliwatt: Generate a Constant 1000Hz tone at 0dbm (mu-law) MixMonitor: Record a call and mix the audio during the recording Monitor: Monitor a channel MP3Player: Play an MP3 file or stream MusicOnHold: Play Music On Hold indefinitely
Re: [Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ Is MeetMe listed if you type show applications in the CLI? No Try specifying a room number in extensions.conf. Does it work without the dynamic stuff? MeetMe([confno][,[options][,pin]]) doesn't work either, I'm guessing because of the above. ___ --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
Re: [Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ From the wiki: If you compiled Asterisk yourself from source and neither /usr/include/linux/zaptel.h nor /usr/local/include/zaptel.h were present (this will be the case if you add the Zaptel drivers after the fact.), certain applications (fzapras, meetme, flash, zapbarge, zapscan and page) will not have been compiled. Ok well that makes sense I'll have to go, and try that again. I thought the whole zap* stuff was just for people using the special hardware. I'll have to go back through, and fix all that. Not a bad time to upgrade to 1.2.6 I guess. Thanks Miles ___ --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
Re: [Asterisk-Users] How to set busy
When you use groups you shouldn't even execute the dial command, but instead use the busy command. On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote: C F wrote: use groups, check the commands/functions group and checkgroup. I guess I can see how this would be useful, but is there no way to get it to return BUSY in DIALSTATUS var? On 4/9/06, Miles Scruggs [EMAIL PROTECTED] wrote: For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available lines, but if the call was routed via [singletrackmind] dial would return busy if the channel already had one call. Thanks Miles ___ --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
RE: [Asterisk-Users] Psgw
Hi, I have download the uplink and test with skype 1.4 2.0. not lucky to me. Only connect on first call then hang. I need to reboot my windows xp everytime. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Wednesday, March 29, 2006 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Psgw Haven't tried this product myself, but according to their spec it's only 1 call. There's another free SIP-Skype gateway from www.nch.com.au called uplink. http://www.nch.com.au/skypetosip/index.html Giordano Grandis wrote: Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session could I have on a single user ? Thanks all Giordano Thanks This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. --- - ___ --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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ From the wiki: If you compiled Asterisk yourself from source and neither /usr/include/linux/zaptel.h nor /usr/local/include/zaptel.h were present (this will be the case if you add the Zaptel drivers after the fact.), certain applications (fzapras, meetme, flash, zapbarge, zapscan and page) will not have been compiled. Ok works like a charm now. So now when I dial ext , I get this: Created MeetMe conference 1023 for conference '0' So my question would be, how do I get other people to join this conference? The voice prompts only tell me that You are entering conference number X where X is 0,1,2 What is the other number I see in the logs 1023 it seems to count down from there for every consecutive conference? Thanks Miles ___ --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
Re: [Asterisk-Users] CallerID
Michelle, you sent a single message containing suggestions to me on 11/02/2005. Your claim to have contacted me many times is clearly false. Due to demands outside the asterisk world, I have not been monitoring the list, but I doubt that should have been necessary, considering that contact information and even a mailing list are available for cid-rewrite. Nobody at all contacted me about reverse lookup not working, and since the script has published was in production here for over a month as well as on many other servers, I have to question the validity of that claim. My comments about spelling and commercial use are very productive. Much unlike you seem to, I take pride in the work I do, and being associated with something so poorly written as your changes to the readme is an embarrassment to both of parties. Additionally, programming is a very exact process, and the quality of your documentation betrays your ability. I do maintain that you are in fact misleading potential downloaders on the origin of the script. You have removed contact information and effectively taken credit for this work. You furthermore are offering paid support, which qualifies as commercial use and you have neither asked for nor been granted permission for commercial use of my intellectual property. Expect no help or cooperation from me in integrating your changes -- your changes are hacks at best, and a far cry from the properly architected changes I have planned and partially integrated in my production script. In the meantime, either remove the download of this bastardized script from your site, or add full contact information back into the readme file and offer FREE support for it. Please comply within 72 hours of receipt of this message. Regards, -- Jay Milk Technical Support wrote: Jay, I contacted you many times regarding the script, whether you planned to update it, suggestions for features, etc. You did not respond to any of my later emails. Similarly, there was discussion between list members regarding whether this script was orphaned after changes to 411.com made the reverse lookup non-functional - for a long time. I assumed responsibility for updating the script as a courtesy to Asterisk users. Your comments about spelling, resale, etc. are abrasive, unproductive, and misleading. Not only is the script available without charge on the web site, credit to you remains with the script - in fact even the download link of the web site gives you credit! And of course, why would I update the script and then encourage users to download an older version from another site? If you have time to dedicate to the cid_rewrite project terrific - I would rather see one stream benefit all users. Let's work to integrate changes going forward. If you would prefer not to, I would be pleased to rename the script so that there is no confusion. Regards, Michelle -Original Message- From: Jay Milk [mailto:[EMAIL PROTECTED] Sent: Saturday, April 08, 2006 1:05 AM To: Technical Support; Asterisk Users Mailing List - Non-Commercial Discussion; Michael Stahl Subject: Re: [Asterisk-Users] CallerID Michelle, 1. Courtesy would suggest that you would have contacted the author of the script (me) to ask permission to modify this and host it elsewhere. 2. What possessed you to remove ALL credits and original download location from the readme file? Are you trying to pawn other people's work off as yours? 3. It's not exactly smart to continue someone else's versioning scheme if you're intending to make a fork. 4. Your spelling is atrocious. 5. The script is not orphaned, even though you seem to imply this in the readme file. Since you are selling support for this script, that qualifies as commercial use and is expressly prohibited by the micro-license included in the original script. Please remove it from your download page until you have made arrangements for further distribution with me. I'm utterly amazed at the bad form I see here. Downloads of the original script are available here: http://www.muware.com/asterisk/ The script is alive and working well, and I've made various enhancements to user-requests in the recent past. -- JM Technical Support wrote: Miles, You can also download cid_rewrite from www.generationd.com This PHP script looks up the phone numbers in a local MySQL table, and/or uses reverse 411 on the web to lookup the name, and/or more options. Michelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Friday, April 07, 2006 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID 2006/4/7, Miles Scruggs [EMAIL PROTECTED]: Could you give me an example code of how this would work, and how to setup the database, I'm pretty new and while what you have written makes sense, and sounds like a good plan I'm not sure I can implement
R: [Asterisk-Users] Psgw
Thanks Gavin. On Uplink i have another kind of problem: the signalling is ok, but when i try to answer my skype give me an error on audio part. It could depend of nat or justabout port not open on my firewall? Thanks Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di kevin ling Inviato: domenica 9 aprile 2006 19.29 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] Psgw Hi, I have download the uplink and test with skype 1.4 2.0. not lucky to me. Only connect on first call then hang. I need to reboot my windows xp everytime. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Wednesday, March 29, 2006 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Psgw Haven't tried this product myself, but according to their spec it's only 1 call. There's another free SIP-Skype gateway from www.nch.com.au called uplink. http://www.nch.com.au/skypetosip/index.html Giordano Grandis wrote: Hi all, anyone never used PSGW as gateway beeween * and SkyPe? If yes, how does it works? How many session could I have on a single user ? Thanks all Giordano Thanks This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. --- - ___ --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 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
Re: [Asterisk-Users] Psgw
kevin ling wrote: Hi, I have download the uplink and test with skype 1.4 2.0. not lucky to me. Only connect on first call then hang. I need to reboot my windows xp everytime. Skype is evil. I would recommend you find a way to spend your time more productively. B. ___ --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
Re: [Asterisk-Users] GXP-2000 and Voicemail
Right, but it's asking for a user id not a number to dial. So, how would I set it to dial extension ? Thanks, Waldo On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote: Look at the Account Settings for Voice Mail UserID. Hi, I have a few GXP-2000 working fine with Asterisk. The one thing I have not been able to do is to program the MSG button to dial the Voicemail extension. How can I program that button? I normally use extension for voicemail. Can anyone shed any light? Thanks, Waldo ___ --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
RE: [Asterisk-Users] Force codec
Disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael StrelnikovSent: Saturday, April 08, 2006 7:25 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw.Thanks.Best regards,Michael ___ --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
Re: [Asterisk-Users] CallerID
I just installed the script, it seems to hang while going out to the web. Is there someway to have it run in the background while a background() is playing or something like that? Thanks Miles Jay Milk wrote: Michelle, 1. Courtesy would suggest that you would have contacted the author of the script (me) to ask permission to modify this and host it elsewhere. 2. What possessed you to remove ALL credits and original download location from the readme file? Are you trying to pawn other people's work off as yours? 3. It's not exactly smart to continue someone else's versioning scheme if you're intending to make a fork. 4. Your spelling is atrocious. 5. The script is not orphaned, even though you seem to imply this in the readme file. Since you are selling support for this script, that qualifies as commercial use and is expressly prohibited by the micro-license included in the original script. Please remove it from your download page until you have made arrangements for further distribution with me. I'm utterly amazed at the bad form I see here. Downloads of the original script are available here: http://www.muware.com/asterisk/ The script is alive and working well, and I've made various enhancements to user-requests in the recent past. ___ --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
Re: [Asterisk-Users] GXP-2000 and Voicemail
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten = 100,4,Macro(hangupcall)
so the user doesn't need to put in a password when they press the MSG button
Waldo Rubinstein wrote:
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can anyone shed any light?
Thanks,
Waldo
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Re: [Asterisk-Users] GXP-2000 and Voicemail
Thanks
Waldo
On Apr 9, 2006, at 2:19 PM, Tim Litwiller wrote:
it dials the userid that you put in that field as an extension.
at home I have it set to 100
and then I have this in the extensions.conf
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,VoicemailMain,s${CALLERIDNUM}
exten = 100,4,Macro(hangupcall)
so the user doesn't need to put in a password when they press the
MSG button
Waldo Rubinstein wrote:
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can anyone shed any light?
Thanks,
Waldo
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Re: [Asterisk-Users] Force codec
Kerry Garrison wrote: Disallow=all allow=ulaw N.B. the problem is depending on extension, not context or protocol. . . B. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Strelnikov *Sent:* Saturday, April 08, 2006 7:25 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw. Thanks. Best regards, Michael 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
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ From the wiki: If you compiled Asterisk yourself from source and neither /usr/include/linux/zaptel.h nor /usr/local/include/zaptel.h were present (this will be the case if you add the Zaptel drivers after the fact.), certain applications (fzapras, meetme, flash, zapbarge, zapscan and page) will not have been compiled. Ok well that makes sense I'll have to go, and try that again. I thought the whole zap* stuff was just for people using the special hardware. I'll have to go back through, and fix all that. Not a bad time to upgrade to 1.2.6 I guess. Thanks Miles ZTDummy=ZapTelDummy, Things like that are always a good time to upgrade. I always do libpri, zaptel, then asterisk, making sure to uncomment the ztdummy entry in the zaptel makefile. Thanks, Steve ___ --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
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, , 2) == Spawn extension (internal, , 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extensionhl=engl=usct=clnkcd=4lr=lang_enclient=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles If you type modprobe zaptel modprobe ztdummy at the Linux CLI, what do you get? Nothing, they were loaded before, and loaded just fine. lsmod Module Size Used by ztdummy 2608 - rtc10620 - zaptel186468 - crc_ccitt 1576 - 3c59x 40240 - ___ And your dialplan for extension ? In extensions.conf exten = ,1, Wait(1) exten = ,2,MeetMe(|Mde) in meetme.conf [rooms] conf = 901 conf = 902 conf = 903 ___ From the wiki: If you compiled Asterisk yourself from source and neither /usr/include/linux/zaptel.h nor /usr/local/include/zaptel.h were present (this will be the case if you add the Zaptel drivers after the fact.), certain applications (fzapras, meetme, flash, zapbarge, zapscan and page) will not have been compiled. Ok works like a charm now. So now when I dial ext , I get this: Created MeetMe conference 1023 for conference '0' So my question would be, how do I get other people to join this conference? The voice prompts only tell me that You are entering conference number X where X is 0,1,2 What is the other number I see in the logs 1023 it seems to count down from there for every consecutive conference? Thanks Miles I am not sure about these newfangled dynamic conferences ;-) I guess you could try dialing from another phone and to dial either 1023 or 0, my guess is 1023 is what the other people will have to dial. Thanks, Steve Totaro ___ --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
Re: [Asterisk-Users] CallerID
Jay Milk wrote: Michelle, you sent a single message containing suggestions to me on 11/02/2005. Your claim to have contacted me many times is clearly false. Due to demands outside the asterisk world, I have not been monitoring the list, but I doubt that should have been necessary, considering that contact information and even a mailing list are available for cid-rewrite. Nobody at all contacted me about reverse lookup not working, and since the script has published was in production here for over a month as well as on many other servers, I have to question the validity of that claim. My comments about spelling and commercial use are very productive. Much unlike you seem to, I take pride in the work I do, and being associated with something so poorly written as your changes to the readme is an embarrassment to both of parties. Additionally, programming is a very exact process, and the quality of your documentation betrays your ability. I do maintain that you are in fact misleading potential downloaders on the origin of the script. You have removed contact information and effectively taken credit for this work. You furthermore are offering paid support, which qualifies as commercial use and you have neither asked for nor been granted permission for commercial use of my intellectual property. Expect no help or cooperation from me in integrating your changes -- your changes are hacks at best, and a far cry from the properly architected changes I have planned and partially integrated in my production script. In the meantime, either remove the download of this bastardized script from your site, or add full contact information back into the readme file and offer FREE support for it. Please comply within 72 hours of receipt of this message. Regards, -- Jay Milk Technical Support wrote: Jay, I contacted you many times regarding the script, whether you planned to update it, suggestions for features, etc. You did not respond to any of my later emails. Similarly, there was discussion between list members regarding whether this script was orphaned after changes to 411.com made the reverse lookup non-functional - for a long time. I assumed responsibility for updating the script as a courtesy to Asterisk users. Your comments about spelling, resale, etc. are abrasive, unproductive, and misleading. Not only is the script available without charge on the web site, credit to you remains with the script - in fact even the download link of the web site gives you credit! And of course, why would I update the script and then encourage users to download an older version from another site? If you have time to dedicate to the cid_rewrite project terrific - I would rather see one stream benefit all users. Let's work to integrate changes going forward. If you would prefer not to, I would be pleased to rename the script so that there is no confusion. Regards, Michelle -Original Message- From: Jay Milk [mailto:[EMAIL PROTECTED] Sent: Saturday, April 08, 2006 1:05 AM To: Technical Support; Asterisk Users Mailing List - Non-Commercial Discussion; Michael Stahl Subject: Re: [Asterisk-Users] CallerID Michelle, 1. Courtesy would suggest that you would have contacted the author of the script (me) to ask permission to modify this and host it elsewhere. 2. What possessed you to remove ALL credits and original download location from the readme file? Are you trying to pawn other people's work off as yours? 3. It's not exactly smart to continue someone else's versioning scheme if you're intending to make a fork. 4. Your spelling is atrocious. 5. The script is not orphaned, even though you seem to imply this in the readme file. Since you are selling support for this script, that qualifies as commercial use and is expressly prohibited by the micro-license included in the original script. Please remove it from your download page until you have made arrangements for further distribution with me. I'm utterly amazed at the bad form I see here. Downloads of the original script are available here: http://www.muware.com/asterisk/ The script is alive and working well, and I've made various enhancements to user-requests in the recent past. -- JM Technical Support wrote: Miles, You can also download cid_rewrite from www.generationd.com This PHP script looks up the phone numbers in a local MySQL table, and/or uses reverse 411 on the web to lookup the name, and/or more options. Michelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Friday, April 07, 2006 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID 2006/4/7, Miles Scruggs [EMAIL PROTECTED]: Could you give me an example code of how this would work, and how to setup the database, I'm pretty new and while what you have written makes sense, and sounds like a good
Re: [Asterisk-Users] 407 proxy authentication
in the sip.conf insecure=very canreinvite=yes []'s - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Sent: Saturday, April 08, 2006 11:41 Subject: [Asterisk-Users] 407 proxy authentication Hello, look at this I can't receive calls from other domains I wish sip:[EMAIL PROTECTED] are forwarded to asterisk however this one spend its time to ask 407 proxy authentication. asterisk 1.2.5 + realtime how can i fix this problem what' wrong ? extension.conf [info] exten = info,1,Answer() exten = info,2,Dial(Sip/84,10) exten = info,3,Dial(Sip/85,10) exten = info,4,Hangup serveur1*CLI sip show user info load serveur1*CLI * Name : info Secret : Not set MD5Secret: Not set Context : info Language : fr AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : ACL : No Codec Order : (g729|ilbc|gsm|ulaw|alaw) harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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
Re: [Asterisk-Users] meetme
Ok works like a charm now. So now when I dial ext , I get this: Created MeetMe conference 1023 for conference '0' So my question would be, how do I get other people to join this conference? The voice prompts only tell me that You are entering conference number X where X is 0,1,2 What is the other number I see in the logs 1023 it seems to count down from there for every consecutive conference? I am not sure about these newfangled dynamic conferences ;-) I guess you could try dialing from another phone and to dial either 1023 or 0, my guess is 1023 is what the other people will have to dial. I would assume that it would work like that, but nope. from a different phone just creates a new conf, and 1023 is never announced it is only in in the logs. where would a person find out about how meetme() works? ___ --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
[Asterisk-Users] How to avoid Avoiding deadlock...
An Asterisk box at customer site shows these messages pretty regularly. This causes one way voice, the called party cannot hear the calling party. Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for '0x817b790', 10 retries! Apr 7 14:47:46 WARNING[18406] channel.c: Avoided initial deadlock for '0x81a4380', 10 retries! Apr 7 14:58:53 WARNING[18406] channel.c: Avoided initial deadlock for '0x817b790', 10 retries! This customer is running Cisco phones, and an IAX trunk top our PSTN GW. The PSTN GW does not show any problems at all. I would appreciate any input regarding this issue. I've seen several posts recently, but no one seems to say what the root cause of this problem is. Thanks in advance, Joe ___ --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
RE: [Asterisk-Users] meetme
Snip.. you could try dialing from another phone and to dial either 1023 or 0, my guess is 1023 is what the other people will have to dial. I would assume that it would work like that, but nope. from a different phone just creates a new conf, and 1023 is never announced it is only in in the logs. where would a person find out about how meetme() works? From the CLI you can do show application meetme. I would suggest you add a conference number to the extension. IE Exten = 750,1,MeetMe(750| your options go here ) Exten = 751,1,MeetMe(750| your options go here ) Exten = 752,1,MeetMe(750| your options go here ) Exten = 753,1,MeetMe(750| your options go here ) That will give you 4 meetme rooms. Play with that and then add and remove the different options that are in MeetMe to 'tweak' your install. Alex ___ --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
Re: [Asterisk-Users] meetme
Snip.. you could try dialing from another phone and to dial either 1023 or 0, my guess is 1023 is what the other people will have to dial. I would assume that it would work like that, but nope. from a different phone just creates a new conf, and 1023 is never announced it is only in in the logs. where would a person find out about how meetme() works? From the CLI you can do show application meetme. I would suggest you add a conference number to the extension. IE Exten = 750,1,MeetMe(750| your options go here ) Exten = 751,1,MeetMe(750| your options go here ) Exten = 752,1,MeetMe(750| your options go here ) Exten = 753,1,MeetMe(750| your options go here ) Yes standard conferences are well described in the wiki, and I have them working fine. unfortunately I want to use the dynamic conferences. 'show application meetme' doesn't tell us anything about how to use the options once you have initialized the application. ___ --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
[Asterisk-Users] Cisco 7960 problems
Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before? Thanks Paul ___ --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
[Asterisk-Users] Problem - Voicemail resets phone
Hi Everyone, Things seem to work fine (except my phone audio issue in a previous mail) I can leave a vmail message and it emails it out fine. However when I dial the vmail server from any phone it usually resets the phone half way through. There is no single point where it starts to do this, it can vary but it happens sometimes after I connect to the vmail server. Has anyone seen this? What other details can I post? Thanks Paul ___ --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
Re: [Asterisk-Users] Cisco 7960 problems
Paul A Brown wrote: Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before? What firmware are you using with the phone? SIP or SCCP? I have 2 7960's with 7914's attached using the latest chan_sccp and have not problems like describe. Regards Jon ___ --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
Re: [Asterisk-Users] question about DISA
hi Ronald, i would use a CallerIDNum authentication, based on the Asterisk Database to solve it. then you do not need any verification. you just build a list of approved numbers in the database and then have a context checking the whitelist. if you need more help, let me know, Mickey On 4/8/06, Ronaldo Chan [EMAIL PROTECTED] wrote: Lists,Hi, good day, i was being task to create a DISA access for internalpurpose of the company, i'm having a problem to work with it with authentication, but i think it's really a straight forward thing to do,can someone enlight me on this. thankssample code snippetexten = 5,Goto(inward,s,1)[inward]exten = s,1,Disa(1234|outgoing) ; DISA appssupposed to ask me a password but it's notinstead it's drop me immedietly to a dial toneexten = s,2,HangupMy Workaround.exten = s,1,Authenticate(1234)exten = s,2,Disa(no-password|outgoing) ThanksRonald___--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
RE: [Asterisk-Users] CallerID
Miles, I think this is a limitation of the AGI - I don't believe that asterisk can fork a new process. If so, that would be interesting! The script uses Wget - I believe we can set a timeout so that your system doesn't hang waiting for the HTTP response. Let me know if that would solve your problem. (You can also set the WGET timeout in your system's config I believe) MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs Sent: Sunday, April 09, 2006 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID I just installed the script, it seems to hang while going out to the web. Is there someway to have it run in the background while a background() is playing or something like that? Thanks Miles ___ --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
Re: [Asterisk-Users] Cisco 7960 problems
In both SCCP and SIP loads, dialtone comes from the phone itself, so if you're not getting that, it's probably a firmware problem. Does it affect all three sound systems (speaker, headset, handset) or just one of them?--JW- Original Message From: Paul A Brown [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Sunday, April 9, 2006 12:41:26 PMSubject: [Asterisk-Users] Cisco 7960 problemsHi All,Not sure if this is a phone problem or an Asterisk problem.Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is.if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before?ThanksPaul ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] Cisco 7960 problems
Hi Its a SIP image, fairly old but seemed to of been ok in the past (I trashed asterisk a while back and recently rebuilt it) P003-07-3-00 I tried upgrading ot the latest but my dialplan.xml didn't work anymore Thanks Paul - Original Message - From: Jon Farmer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 09, 2006 8:49 PM Subject: Re: [Asterisk-Users] Cisco 7960 problems Paul A Brown wrote: Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before? What firmware are you using with the phone? SIP or SCCP? I have 2 7960's with 7914's attached using the latest chan_sccp and have not problems like describe. Regards Jon ___ --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
Re: [Asterisk-Users] Cisco 7960 problems
Do you have a sccp config example I could look at Thanks - Original Message - From: Jon Farmer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 09, 2006 8:49 PM Subject: Re: [Asterisk-Users] Cisco 7960 problems Paul A Brown wrote: Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a dialtone. After around 30 minutes and I pick up phone I get no dial tone but I can still dial. I dialled the voicemail number, I can see on the asterisk console its asking for which vmail box and password but I hear nothing. Anyone heard anything like this before? What firmware are you using with the phone? SIP or SCCP? I have 2 7960's with 7914's attached using the latest chan_sccp and have not problems like describe. Regards Jon ___ --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
Re: [Asterisk-Users] Cisco 7960 problems
Paul A Brown wrote: Do you have a sccp config example I could look at http://www.voip-info.org/wiki/view/SCCP-HOWTO2 Regards Jon ___ --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
[Asterisk-Users] Re: oh323.conf problem
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I have had the exact same problem last week. I have not yet solved it. So instead I am using ooh323, but would prefer to use oh323. Can anyone help? I'm glad that I'm not the only one :)) Hopefully we'll find solution to this problem. -- Tomislav Parcina tparcina#lama.hr ___ --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
Re: [Asterisk-Users] Force codec
What about different extensions using different connections? Paul Hales Technical Manager AsteriskIT - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 10, 2006 4:26 AM Subject: Re: [Asterisk-Users] Force codec Kerry Garrison wrote: Disallow=all allow=ulaw N.B. the problem is depending on extension, not context or protocol. . . B. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Michael Strelnikov *Sent:* Saturday, April 08, 2006 7:25 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw. Thanks. Best regards, Michael 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme
Miles Scruggs wrote: Ok works like a charm now. So now when I dial ext , I get this: Created MeetMe conference 1023 for conference '0' So my question would be, how do I get other people to join this conference? The voice prompts only tell me that You are entering conference number X where X is 0,1,2 What is the other number I see in the logs 1023 it seems to count down from there for every consecutive conference? I am not sure about these newfangled dynamic conferences ;-) I guess you could try dialing from another phone and to dial either 1023 or 0, my guess is 1023 is what the other people will have to dial. I would assume that it would work like that, but nope. from a different phone just creates a new conf, and 1023 is never announced it is only in in the logs. where would a person find out about how meetme() works? ___ Google is always my first stop. Try googling asterisk pbx meetme or go to www.voip-info.org, which is where google will most likely point you to anyways. ___ --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
[Asterisk-Users] Provisioning Server...
Hello Everyone. I have a question, I have set up [EMAIL PROTECTED] and I have purchased an ATA from TigerDirect. It is an MTA-102 from VOIP Solutions, when I check the website that is in the documentation it is in Portugese from Brazil. I have done some investigation and it seems that the MTA is provisioned through a provisioning server. Does anyone know how to configure a provisioning server through Asterisk? Is it worth it? The MTA-102 was real cheap, like me, so I was hoping to use it. Anyone have any suggestions??? Thanks in advance. Andrew Saving just one dog will not change the world, but it will change the world for that one dog. ___ --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
Re: [Asterisk-Users] ANI on a PRI
Hate to reply to my own posting but I wonder if anyone know the answer? Steve Totaro wrote: Is there a setting somewhere in * to define whether I am receiving callerID or true ANI? Global Crossing claims they are sending ANI but I dont think so. My understanding of ANI is that it is always sent, regardless if callerID is blocked. If I dial *67 and my DID, I get Presentation: Presentation prohibited of network provided number and no number. Before I call GC on Monday to complain, I want to make sure I am correct in my understanding of ANI and not missing something on my side. Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 04 e1 81 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 02 21 a3] Calling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation prohibited of network provided number (35) '' ] [70 0b a1 38 30 30 35 36 34 30 38 31 39] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8005640819' ] -- Making new call for cr 16 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: CALL PROCEEDING (2) [18 04 e9 81 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Protocol Discriminator: Q.931 (8) len=15 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: CONNECT (7) [18 04 e9 81 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: CONNECT ACKNOWLEDGE (15) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 16/0x10) (Terminator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 16/0x10) (Originator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ --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
[Asterisk-Users] MWI Problem
When I register to a remote Asterisk system using IAX2, I can see it notifying my Asterisk box that I have voicemail waiting. How can I get Asterisk to use that information and send WMI to one or more of my SIP phones? Thanks, Eric ___ --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
Re: [Asterisk-Users] Force codec
I want to make it global.On 4/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:What about different extensions using different connections? Paul HalesTechnical ManagerAsteriskIT- Original Message -From: Brian Capouch [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, April 10, 2006 4:26 AMSubject: Re: [Asterisk-Users] Force codec Kerry Garrison wrote: Disallow=all allow=ulaw N.B. the problem is depending on extension, not context or protocol. . . B. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] *On Behalf Of *Michael Strelnikov *Sent:* Saturday, April 08, 2006 7:25 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Force codec Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw. Thanks. Best regards, Michael 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 Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards,Michael Strelnikov ___ --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
[Asterisk-Users] for review
I made an edit to the wiki: http://www.voip-info.org/wiki/view/Asterisk+tips+campon While I need this solution, and I think that some other people can benefit from various aspects of it, can anyone see if there is a more elegant solution to achieve the same result? Please feel free to edit that example so it is the best for everyone, I'm a true amateur, and I think that example needs a lot of help. Thanks Miles ___ --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
Re: [Asterisk-Users] Force codec
I meant dialled extension, not originating
extension.
like :
exten =
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Michael
Strelnikov
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 11:13
AM
Subject: Re: [Asterisk-Users] Force
codec
I want to make it global.
On 4/10/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:
What
about different extensions using different connections? Paul
HalesTechnical ManagerAsteriskIT- Original Message
-From: "Brian Capouch" [EMAIL PROTECTED]To: "Asterisk
Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comSent:
Monday, April 10, 2006 4:26 AMSubject: Re: [Asterisk-Users] Force
codec Kerry Garrison wrote:
Disallow=all allow=ulaw N.B. the
problem is "depending on extension," not context or protocol. .
. B.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] *On Behalf Of *Michael
Strelnikov *Sent:* Saturday, April 08,
2006 7:25 PM *To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Force
codec Hi,
Is it
possible to force using codec depends on extension? For
example, voice codec is ILBC and with some
prefix fax code should be
ulaw. Thanks.
Best regards,
Michael http://lists.digium.com/mailman/listinfo/asterisk-users
___
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-- Best
regards,Michael Strelnikov
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Re: [Asterisk-Users] wellgate registration 3802
There is two entry on the username which is the number assigned to the port and the username apper on the sip entry u should put the same on to it.. The number assinged is the authuser Jerry Geis wrote: I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend dtmfmode=inband username=1002 secret=1002 host=dynamic canreinvite=yes nat=no context=wellgate Apr 7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.24' - Username/auth name mismatch Apr 7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.24' - Username/auth name mismatch I am getting these two errors on the console. What have I missed that will let the wellgate 3802 connect to asterisk? Thanks, jerry ___ --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 begin:vcard fn:Jolly M. Recto n:Recto;Jolly org:Telekoms Philippines Inc.;Satellite-Engineering adr:San Juan;;168 Luna Mencias Street;Manila;Manila;1500;Philippines email;internet:[EMAIL PROTECTED] title:Manager tel;work:+632-7268686 tel;fax:+632-7241916 tel;cell:+63-9196611066 x-mozilla-html:FALSE url:http://www.itextron.com version:2.1 end:vcard ___ --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
[Asterisk-Users] Instant Message?
Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any relateddocuments or weblinks?-- Thanks Best Regards! Steven Li ___ --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
RE: [Asterisk-Users] Uplink Skype2Sip
In my remember. The uplink install a virtual sound card. So uplink can auto answer the call from skype or sip side and redirect to another side. No matter what kind of onboard audio card do you have. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Saturday, April 08, 2006 2:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uplink Skype2Sip I cant make the proggie link my sip to skype, but skype to sip work great. Im running winxpsp2 with a cheapo onboard sound card. On 4/7/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone get it worked ? Uplink route me the call incoming from skype but when i answer, my skype go in error on sound card ? I also set in my hosts this value: 127.0.0.1 pgp01.televolution.net 127.0.0.1 stun01.sipphone.com This is my sip.conf [skype] language = it username = skype secret = password host = dynamic defaultip = lan_ip_address_of_uplink port = 5060 type = friend context = from_eth canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 fromuser = skype_username insecure = very qualify = yes callerid = Test 999 allow = all Thanks all Giordano ___ --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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --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
[Asterisk-Users] Manager API Help
Hi All, Could someone send me a code frag on how to get a record from the asterisk database into a PHP variable via the Manager API? I can issue calls, etc. from Manager. But I'm not comprehending how to manipulate database variables. Thanks much. Darren Ellis ___ --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
Re: [Asterisk-Users] Instant Message?
Zhiqiang Li wrote: Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any related documents or weblinks? -- Thanks Best Regards! Steven Li I am not sure exactly what you are trying to do but Jive Messenger has asterisk add-ons and functionality. Might be worth a look for ya. Thanks, Steve Totaro ___ --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
RE: [Asterisk-Users] Instant Message?
I tried the latest version of Jive over the weekend and I have to say it is a giant pile of crap. I did this on multiple machines on both Linux and Windows, and after setting everything up, the moment you add the asterisk module, all authentication and user setup is lost and there is no way to log back in as the admin to fix it. If anyone has any more positive experience I would like to hear about it as it sounds very interesting. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, April 09, 2006 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Instant Message? Zhiqiang Li wrote: Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any related documents or weblinks? -- Thanks Best Regards! Steven Li I am not sure exactly what you are trying to do but Jive Messenger has asterisk add-ons and functionality. Might be worth a look for ya. Thanks, Steve Totaro ___ --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
RE: [Asterisk-Users] Planet VIP-320 DECT gateway with Asterisk?
Hi Sorry for my chinese engish first. The VIP-320 seems like a SIP ATA+DECT Phone product. Please check you have registered to the asterisk server first. Because the VIP-320 built-in H.323/SIP dual mode. The web config tool not so clear. Reference the Page.31. Input the SIP:asterisk_ip_address in the *1 field. And try to sip debug peer and capture the sip message in the CLI mode. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: Thursday, April 06, 2006 11:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Planet VIP-320 DECT gateway with Asterisk? Hello, I just received what seems to be a nice SIP-DECT gateway but can't make it work with asterisk. The manual is very unclear (written in chinese english) and the web configurator is ambiguous as well. Has anyone succeeded in making one of these babies work with * ? info: http://www.planet.com.tw/product/product_dm.php?product_id=367menu_id=3 ___ --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
Re: [Asterisk-Users] Force codec
But in this case you have to define two users on both sides. It is not most likely.On 4/10/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I meant dialled extension, not originating
extension.
like :
exten =
_37X,1,Dial(IAX2/FAX/${EXTEN})
exten =
_38X,1,Dial(IAX2/NOTFAX/${EXTEN})
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
Michael
Strelnikov
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 11:13
AM
Subject: Re: [Asterisk-Users] Force
codec
I want to make it global.
On 4/10/06, [EMAIL PROTECTED]
[EMAIL PROTECTED]
wrote:
What
about different extensions using different connections? Paul
HalesTechnical ManagerAsteriskIT- Original Message
-From: Brian Capouch [EMAIL PROTECTED]To: Asterisk
Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent:
Monday, April 10, 2006 4:26 AMSubject: Re: [Asterisk-Users] Force
codec Kerry Garrison wrote:
Disallow=all allow=ulaw N.B. the
problem is depending on extension, not context or protocol. .
. B.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
] *On Behalf Of *Michael
Strelnikov *Sent:* Saturday, April 08,
2006 7:25 PM *To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Force
codec Hi,
Is it
possible to force using codec depends on extension? For
example, voice codec is ILBC and with some
prefix fax code should be
ulaw. Thanks.
Best regards,
Michael http://lists.digium.com/mailman/listinfo/asterisk-users
___
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-- Best
regards,Michael Strelnikov
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___--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Best regards,Michael Strelnikov
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[Asterisk-Users] PRI Group Calling
Hi,I have a single PRI span setup at present and need to dial a prefix number in order to suppress outgoing caller ID.I am about to have a second PRI Span set up on the same server, but I want to bring both spans into the one group. The second span will be from a different telco. I want to suppress outbound caller ID on this span too, but suspect I will run into the issue of having to dial a different prefix number for this span in order to suppress outbound caller ID.I am sure I'm not the only one out there that has had this problem. How do I get around the issue? Do I need to detect the outbound span before dialing on the group, or do I need to separate the two spans into different groups and fail-over manually using a chanavail strategy? Ideally I'd like to just issue a single Dial(Zap/g2/prefixnumber) but I am sure my second telco will come back with a different prefix than the other.Any hints gratefull received.cheers, Mark.http://www.switchnet.com.auhttp://www.hearmymessage.com ___ --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
RE: [Asterisk-Users] Instant Message?
I have Jive (wildfire) 2.4.4 running on a win2k3 server box with the asterisk plugin. Installed without significant problems, has been up and running for about 6 wks now. Conference rooms especially are convenient. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Sunday, April 09, 2006 7:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Instant Message? I tried the latest version of Jive over the weekend and I have to say it is a giant pile of crap. I did this on multiple machines on both Linux and Windows, and after setting everything up, the moment you add the asterisk module, all authentication and user setup is lost and there is no way to log back in as the admin to fix it. If anyone has any more positive experience I would like to hear about it as it sounds very interesting. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, April 09, 2006 6:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Instant Message? Zhiqiang Li wrote: Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any related documents or weblinks? -- Thanks Best Regards! Steven Li I am not sure exactly what you are trying to do but Jive Messenger has asterisk add-ons and functionality. Might be worth a look for ya. Thanks, Steve Totaro ___ --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 This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --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
Re: [Asterisk-Users] PRI Group Calling
Mark Edwards wrote: I have a single PRI span setup at present and need to dial a prefix number in order to suppress outgoing caller ID. Really? Normally you would set the calling presentation to 'restricted' on a PRI, no prefix would be needed. ___ --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
[Asterisk-Users] Asterisk Dial Command Timeout not Accurate (not even close)
I have an issue with trying to ensure that when dialling an extension that it continues to ring up to the timeout value. But what I am finding is that the timeout is all over the place. Sometimes half the timeout value and other times within a few seconds of the timeout value. I am running with the following software versions on a Dell PE2550 dual-processor server that is not under load; Red Hat FC4 Kernel 2.6.15-1.1830_FC4smp Asterisk Version 1.2.6 Zaptel Version 1.2.5 Libpri Version 1.2.2 Addons Version 1.2.2 Sounds Version 1.2.1 How do I ensure that the dial command timeout is somewhat more accurate. ___ --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
Re: [Asterisk-Users] wellgate registration 3802
10 apr 2006 kl. 04.02 skrev Jolly M. Recto: There is two entry on the username which is the number assigned to the port and the username apper on the sip entry u should put the same on to it.. The number assinged is the authuser The username setting is not needed in this configuration at all. Jerry Geis wrote: I have a new wellgate 3802 unit. I have not gotten it to register with asterisk 1.2.6. My proxy setting is the correct IP in the 3802. My security config is 1001/1001 and 1002/1002 on the wellgate (simple at this time). My sip.conf has: [wellgate3802L1] type=friend dtmfmode=inband username=1001 secret=1001 host=dynamic canreinvite=yes nat=no context=wellgate [wellgate3802L2] type=friend dtmfmode=inband username=1002 secret=1002 host=dynamic canreinvite=yes nat=no context=wellgate Apr 7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: Registration from 'sip: [EMAIL PROTECTED]' failed for '192.168.1.24' - Username/auth name mismatch Apr 7 11:54:47 NOTICE[6288]: chan_sip.c:10879 handle_request_register: Registration from 'sip: [EMAIL PROTECTED]' failed for '192.168.1.24' - Username/auth name mismatch I am getting these two errors on the console. What have I missed that will let the wellgate 3802 connect to asterisk? You have no account named 1001. Your account names are wellgate3802L1 and wellgate3802L2 in this configuration. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk European Tour: http:// www.meetasterisk.com - Register today! * Asterisk Training http://edvina.net/training/ ___ --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
