Re: [Asterisk-Users] '.' pattern and non-SIP phones
On Thu, 25 Sep 2003, Andrew Kohlsmith wrote:
Using FWD and accessing it via this extension:
exten = _*8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This works *perfectly* with SIP phones. However with a regular phone
plugged into an FXS card (PhoneJack PCI in my case) the '.' traps the first
number dialled after *8 and tries calling that. I've tried setting a digit
timeout but it doesn't seem to help.
Changing that to
exten = _*8X,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
works, but is hardly optimal, since I plan on changing my dialplan to allow
varied-length numbers for other things.
I can't explain it without looking at the code, and I'm short on time so I
won't go there but the way that works best for me is:
exten = _*8X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
James
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Re: [Asterisk-Users] G729 experiences..
Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong Another question.. Is zaptel hardware required in order to use the G.729 codec?? The reason for the, what may seem like a silly question is that on the digium website they comment The G.729 codec works with all Digium cards.. I am wondering what relationship there is between digium cards and codecs?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound file script
Logged in this morning to find that the sound file scripts are now up on the Wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files Courtesy of Zac Sprackett, I believe. Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
Can't really remember, If I am not mistaken you dont have to reregister the codec. unless you format your harddisk. If your using chan_h323, you need to modify its makefile to compile with g.729 support every time you download from cvs.(something that I always forgot to do) :). Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:37 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
No, I dont think you need a zap device. I used to run meetme, where all conference participants are from IP endpoints (G.729) without any zaptel device. I just added a digium E100P recently, works without problem so far. I am not sure about the relationship, may be what they mean is IP endpoints callling PSTN lines through asterisk(with zap devices) works using digium's G.729. Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:43 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong Another question.. Is zaptel hardware required in order to use the G.729 codec?? The reason for the, what may seem like a silly question is that on the digium website they comment The G.729 codec works with all Digium cards.. I am wondering what relationship there is between digium cards and codecs?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX calling number
Hello, I am recentlyinspecting the IAX protocol.. I wonder if there away to associate a user name to a number say I have a client register to the IAX server with username 'John' andI want to associate a number say '12345678' tho John so otherregister users can call john by dialing 12345678. Something like the H323_id and the E164 alias in H323 protocol. Foong
Re: [Asterisk-Users] IAX calling number
On Fri, 26 Sep 2003, Chee Foong wrote: I wonder if there away to associate a user name to a number say I have a client register to the IAX server with username 'John' and I want to associate a number say '12345678' tho John so other register users can call john by dialing 12345678. Something like the H323_id and the E164 alias in H323 protocol. exten = 12345768,1,Dial(IAX/john) - wasim of the it doesn't get any simpler than this cult ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX calling number
Ahh...Understood. That's possible.
But my problem is I will have 500 users (and increasing). I can't have an
entry for every users in the config file. The only way to handle this so far
I found is to use number as username, therefore we can use only 1 extension:
exten = _700XX,1,Dial(IAX/${EXTEN})
But user wont like it if username is a long string of number, they prefer
meaningful name.
Thanks anyway.
Foong
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 26, 2003 3:47 PM
Subject: Re: [Asterisk-Users] IAX calling number
On Fri, 26 Sep 2003, Chee Foong wrote:
I wonder if there away to associate a user name to a number say I have a
client register to the IAX server with username 'John' and I want to
associate a number say '12345678' tho John so other register users can
call john by dialing 12345678. Something like the H323_id and the E164
alias in H323 protocol.
exten = 12345768,1,Dial(IAX/john)
- wasim of the it doesn't get any simpler than this cult
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Re: [Asterisk-Users] Does SIP work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 25 September 2003 20:35, Olle E. Johansson wrote: I know, I just meant that pretty much everything else is either descriptive or described in sip.conf. Except the meaning of [xxx] entries. http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+channels Do you think we've missed your point there? If so, help us correct this problem. All pages in the wiki are open for editing. Nice page. I've added a comment. Don't know if my explanation is any good though. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/dADG2TEAILET3McRAodmAKCSXZcEJQ74Hk1bkesbddsv6BJODQCgh1Yk WDXkixP0dFhyI5RQVOnyhEw= =7e9D -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP routing..
Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
Matthew Hardeman wrote: It's ok... The voice sounds fine. It's superior to most cell phone calls, anyway. I've used it with the Cisco 7960's without any trouble. You can use asterisk in any way that uses it in console mode. Safe asterisk does so, so you can use it. This may be otherwise fixed, but I'm not sure. As safe asterisk works, I don't worry about it. Voicemail will use one license for each output stream it has to transcode. Therefore, it is preferable if you are using G729 to only write out one format of voicemail recording. I use WAV49, which is small like GSM, but easier to play on default windows installs with any kind of decent media player installed. It *does* properly release the license when done. (At least now, on my system, it does.) Matt Hardeman PaperSoft Has anyone used the Digim G.729 codec with SNOM 200 phones? I have heard people have had success with the Grandstream phones.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
Hi Carl i see web frontend i action is very good!! The total time at end is good thing. Thanks for great work. Can you put the script in some place to download. Dimitri *This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table. If interested, email me on [EMAIL PROTECTED] Do not reply directly to this email, it will bounce. Depending on the level of interest, I may post this somewhere for your free downloading pleasure. Regards, Jamie Carl Jazz Inc. http://www.jazz-inc.net Email: [EMAIL PROTECTED] JID: [EMAIL PROTECTED] Phone: +61-414-365466 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP routing..
WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP routing..
Hi Adam, No queuing won't be an option.. all the traffic I am thinking about will be voice traffic moving in and out of the Asterisk box.. Are you setting up this same senario where you are boing to have two data paths?? Later.. Low, Adam wrote: WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
On Fri, Sep 26, 2003 at 03:54:57PM +0800, Chee Foong wrote: And what I need to do if my asterisk box don't have a harddisk ? I plan to make it on flash or tftpbooting ... May be somebody comment this ? Can't really remember, If I am not mistaken you dont have to reregister the codec. unless you format your harddisk. If your using chan_h323, you need to modify its makefile to compile with g.729 support every time you download from cvs.(something that I always forgot to do) :). Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:37 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ... All opinions expressed are mine and not those of my employer. Yours, Max [Msg N 2278] --- mailto: [EMAIL PROTECTED] phone: +380-44-2054455 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ERROR MESSAGE
Hi! Thaanks the problem was the same, now i´m using a static ip and all is working fine. regards - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 4:07 PM Subject: RE: [Asterisk-Users] ERROR MESSAGE I had this problem when I changed the IP of one of the * boxes. Did not see it on the other boxes. Have you changed the IP of your * box since compiling * first time? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP routing..
Hi, I work for an ISP (c; So I am going to build over the weekend a single Asterisk (RH9) box with two IP addresses (separate subnets) on the same NIC with a L2 ethernet switch connected then two separate routers (one for each subnet) and then try and make some calls to my production Asterisk box. I'll run EtheReal on the same L2 switch so I can see all the SIP and RTP packets so once its setup it should only take a few test calls to figure out exactly whats going on ... Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 13:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP routing.. Hi Adam, No queuing won't be an option.. all the traffic I am thinking about will be voice traffic moving in and out of the Asterisk box.. Are you setting up this same senario where you are boing to have two data paths?? Later.. Low, Adam wrote: WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. - Original Message - From: Keith O'Brien [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 4:29 PM Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK It doesn't matter. The session target in a cisco voip-dial peer always has to point at the far end. In the case of H.323 this would still be the * box. Pointing this at the local router eth0 interface definitely is not correct. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, September 25, 2003 2:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK But I am not using SIP I am using H.323 - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 3:30 PM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK Looks like your sip destination is the same IP as the IP on the ethernet interface. I am pretty sure that this IP needs to be the sip server, not the router. -Sean On Wed, 24 Sep 2003, Bartosz Jozwiak wrote: This is my configuration of my cisco router and still it does not want to work :( Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname asterisk ! aaa new-model aaa authentication login default local enable secret 5 $1$bJzJ$bjJ.hc0TbiopbjjMUnyhg/ ! username admin password 7 07002C494908 ! ! ! ! ip subnet-zero ip name-server 66.178.37.211 ! ! ! ! voice-port 1/0/0 ! voice-port 1/0/1 ! voice-port 1/1/0 ! voice-port 1/1/1 connection plar ! ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ! interface Serial0/0 no ip address no ip directed-broadcast shutdown ! interface Ethernet0/1 no ip address no ip directed-broadcast shutdown half-duplex ! ip classless ip route 0.0.0.0 0.0.0.0 66.178.36.4 no ip http server ! ! line con 0 transport input none line aux 0 line vty 0 4 ! no scheduler allocate end - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 2:25 PM Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080 093f62.shtml That covers the thridparty h323 stuff with * bkw On Wed, 24 Sep 2003, Sean Figgins wrote: That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP routing..
Excellent, I will be very interested in your findings because this is going to be an issue for me in the not too distant future if things go according to plan.. Later.. Low, Adam wrote: Hi, I work for an ISP (c; So I am going to build over the weekend a single Asterisk (RH9) box with two IP addresses (separate subnets) on the same NIC with a L2 ethernet switch connected then two separate routers (one for each subnet) and then try and make some calls to my production Asterisk box. I'll run EtheReal on the same L2 switch so I can see all the SIP and RTP packets so once its setup it should only take a few test calls to figure out exactly whats going on ... Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 13:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP routing.. Hi Adam, No queuing won't be an option.. all the traffic I am thinking about will be voice traffic moving in and out of the Asterisk box.. Are you setting up this same senario where you are boing to have two data paths?? Later.. Low, Adam wrote: WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP codecs Errors
Mark: When I update from CVS and delete allow=all in sip.conf it works great. If allow=all remains in sip.conf it doesn't work. Anyway, It's working again. Thanks a lot! Regards, Gus - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 12:08 AM Subject: Re: [Asterisk-Users] SIP codecs Errors Fixed in CVS On Thu, 25 Sep 2003, CW_ASN wrote: Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The show codecs command shows: *CLI show codecs 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video The sip debug show the following: *CLI sip debug SIP Debugging Enabled Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060 From: 52880472 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown Date: Thu, 25 Sep 2003 16:49:48 ARBUE Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1091135146-4006089175-2409868731-3383986922 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1064519388 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Expires: 180 Content-Type: application/sdp Content-Length: 167 v=0 o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96 s=SIP Call c=IN IP4 172.16.254.96 t=0 0 m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535 15 headers, 6 lines Using latest request as basis request Sending to 172.16.254.96 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format UNKN Capabilities: us - 0, them - 269/0, combined - 0 Non-codec capabilities: us - 1, them - 0, combined - 0 WARNING[1125329600]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060 From: 52880472 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown Date: Thu, 25 Sep 2003 16:49:48 ARBUE Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1091135146-4006089175-2409868731-3383986922 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1064519388 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Expires: 180 Content-Type: application/sdp Content-Length: 167 v=0 o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96 s=SIP Call c=IN IP4 172.16.254.96 t=0 0 m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535 15 headers, 6 lines Ignoring this request Looking for 2060 in default list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.254.96:5060 From: 52880472 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown;tag=as2767183f Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 172.16.254.96:5060 -- Executing VoiceMail(SIP/-0812ba78, u2060) in new stack We're at 172.16.254.96 port 16464 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.254.96:5060 From: 52880472 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown;tag=as2767183f Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 109 v=0 o=root 3781 3781 IN IP4 172.16.254.96 s=session c=IN IP4 172.16.254.96 t=0 0 m=audio 16464 RTP/AVP to 172.16.254.96:5060 == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm-theperson' Sip read: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.254.96:5060 From: 52880472 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown;tag=as2767183f Date: Thu, 25 Sep 2003 16:49:48 ARBUE Call-ID: [EMAIL PROTECTED] User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Max-Forwards: 6 Timestamp: 1064519388 CSeq: 102 BYE Content-Length: 0 11 headers, 0 lines Sending to 172.16.254.96 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP
Re: [Asterisk-Users] SIP codecs Errors
I think there is a problem in CVS because yesterday I updated Asterisk from CVS and I had the same problem with codecs. When I went back with CVS everything was working again normal. -- Bart - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 8:22 PM Subject: Re: [Asterisk-Users] SIP codecs Errors On Thursday 25 September 2003 15:01, CW_ASN wrote: Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The show codecs command shows: *CLI show codecs 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video 'show codecs' in no way, shape, or form indicates what codecs are useable in your sip config. It returns the same output on ALL machines. Look in your sip.conf for what codecs you have available. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Check and restart script..
-Original Message- From: Garry Adkins [mailto:[EMAIL PROTECTED] I agree... As a unix support person at work, I find that I have to write these types of watchdogs often... Sometimes an application will partially fail, or fail but not exit, ending up as some zombie. (I've tried the ps -auxw, and it's not smart enough to see a program has hung... and your load average is now about 80...) A good, no, excellent, monitoring system is Nagios (www.nagios.org). It uses the concepts of plugins to monitor 'OK', 'WARNING' and 'CRITICAL' states. A variety of plugin's for * could monitor the main process, loop back inside of * (via AGI), grep of /var/log/asterisk/messages for channel errors, etc. Nagios can also use event handlers to do things such as restarting processes (*). You're points make sense Garry, and are appreciated. Methods for monitoring the health of * is something to do once I integrate * into our production facility for out-calling of alerts using festival. Until then, I'll rely upon our organic monitoring system, the users. :) Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
Where did you install asterisk? foong - Original Message - From: Max Speransky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 7:07 PM Subject: Re: [Asterisk-Users] G729 experiences.. On Fri, Sep 26, 2003 at 03:54:57PM +0800, Chee Foong wrote: And what I need to do if my asterisk box don't have a harddisk ? I plan to make it on flash or tftpbooting ... May be somebody comment this ? Can't really remember, If I am not mistaken you dont have to reregister the codec. unless you format your harddisk. If your using chan_h323, you need to modify its makefile to compile with g.729 support every time you download from cvs.(something that I always forgot to do) :). Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:37 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ... All opinions expressed are mine and not those of my employer. Yours, Max [Msg N 2278] --- mailto: [EMAIL PROTECTED] phone: +380-44-2054455 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP routing..
WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] RTP routing..
Hi, Sorry for my bad english but I´ll try to explain my problem I got an Asterisk running in my house with ADSL... I´m using X100P and TDM400P cards My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here´s the map with Firewalls Call for anyone to my house = PSTN = X100P = EXTENSIONS = SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE It´s working very nice, but I had to disable iptables in my Asterisk Box(Home)... I was using my linux with PPPoe Client, DynamicDnsClient and IPTABLES... I´d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... Andre Lomonaco -Mensagem original- De: Low, Adam [mailto:[EMAIL PROTECTED] Enviada em: Friday, September 26, 2003 9:06 AM Para: '[EMAIL PROTECTED]' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP routing..
I can restrict the RTP ports used with my Cisco 79xx phones and on my Cisco AS5300 and I think you can with Asterisk by using the rtp.conf but I'm not completely sure, I'd suggest diving into the source for that one ... -Original Message- From: Andre Lomonaco [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 14:31 To: '[EMAIL PROTECTED]' Subject: RES: [Asterisk-Users] RTP routing.. Hi, Sorry for my bad english but I´ll try to explain my problem I got an Asterisk running in my house with ADSL... I´m using X100P and TDM400P cards My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here´s the map with Firewalls Call for anyone to my house = PSTN = X100P = EXTENSIONS = SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE It´s working very nice, but I had to disable iptables in my Asterisk Box(Home)... I was using my linux with PPPoe Client, DynamicDnsClient and IPTABLES... I´d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... Andre Lomonaco -Mensagem original- De: Low, Adam [mailto:[EMAIL PROTECTED] Enviada em: Friday, September 26, 2003 9:06 AM Para: '[EMAIL PROTECTED]' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie: Crossing my fingers
I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a document to setup as well as the API calls (and a book). Anything like this for Asterisk? Thanks for your patience. -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RES: [Asterisk-Users] RTP routing..
On Fri, 2003-09-26 at 14:30, Andre Lomonaco wrote: My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. rtp.conf When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... because you have no ports open for the RTP, just open those you have defined in rtp.conf -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP routing..
Andre, Yes this is simply controlled using the rtp.conf.. the default is to use UDP ports 1-2.. So if you set IPTABLES to allow inbound traffic to UDP port 5060 and 1-2 your SIP client should work fine.. Later.. Andre Lomonaco wrote: Hi, Sorry for my bad english but I´ll try to explain my problem I got an Asterisk running in my house with ADSL... I´m using X100P and TDM400P cards My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here´s the map with Firewalls Call for anyone to my house = PSTN = X100P = EXTENSIONS = SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE It´s working very nice, but I had to disable iptables in my Asterisk Box(Home)... I was using my linux with PPPoe Client, DynamicDnsClient and IPTABLES... I´d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... Andre Lomonaco -Mensagem original- De: Low, Adam [mailto:[EMAIL PROTECTED] Enviada em: Friday, September 26, 2003 9:06 AM Para: '[EMAIL PROTECTED]' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
costas wrote: I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a document to setup as well as the API calls (and a book). Anything like this for Asterisk? Thanks for your patience. Costas, My install guide may help you most of the way.. http://members.lycos.co.uk/wipe_out/asterisk/ You will have to add the samba and any other packages you need.. Then I would suggest you read the handbooks that are available on the digium.com website on the documentation page.. If you don't have an IP hard phone then you probably want to download X-Lite from www.xten.com That should get you on your way.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP routing..
Yes you can specify which RTP port to use in rtp.conf then you can nicely allow those ports to be open in your iptables. Doing the same thing here myself. Greetings, Tj - Original Message - From: Andre Lomonaco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 2:30 PM Subject: RES: [Asterisk-Users] RTP routing.. Hi, Sorry for my bad english but I´ll try to explain my problem I got an Asterisk running in my house with ADSL... I´m using X100P and TDM400P cards My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here´s the map with Firewalls Call for anyone to my house = PSTN = X100P = EXTENSIONS = SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE It´s working very nice, but I had to disable iptables in my Asterisk Box(Home)... I was using my linux with PPPoe Client, DynamicDnsClient and IPTABLES... I´d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... Andre Lomonaco -Mensagem original- De: Low, Adam [mailto:[EMAIL PROTECTED] Enviada em: Friday, September 26, 2003 9:06 AM Para: '[EMAIL PROTECTED]' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie: Crossing my fingers
Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a document to setup as well as the API calls (and a book). Anything like this for Asterisk? My install guide may help you most of the way.. http://members.lycos.co.uk/wipe_out/asterisk/ You will have to add the samba and any other packages you need.. Then I would suggest you read the handbooks that are available on the digium.com website on the documentation page.. You might also find http://www.automated.it/guidetoasterisk.htm - useful. Matthew -- Outgoing mail is certified Virus Free. Checked by AVG Anti-Virus (http://www.grisoft.com). Version: 7.0.176 / Virus Database: 260.1.3 - Release Date: 23/09/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '.' pattern and non-SIP phones
*8 is call pickup. Can you choose a different extension? If not, I'm
going to have to make call pickup not be checked if you don't have a
pickup/ring group.
Mark
On Fri, 26 Sep 2003, James Golovich wrote:
On Thu, 25 Sep 2003, Andrew Kohlsmith wrote:
Using FWD and accessing it via this extension:
exten = _*8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This works *perfectly* with SIP phones. However with a regular phone
plugged into an FXS card (PhoneJack PCI in my case) the '.' traps the first
number dialled after *8 and tries calling that. I've tried setting a digit
timeout but it doesn't seem to help.
Changing that to
exten = _*8X,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
works, but is hardly optimal, since I plan on changing my dialplan to allow
varied-length numbers for other things.
I can't explain it without looking at the code, and I'm short on time so I
won't go there but the way that works best for me is:
exten = _*8X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
James
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Re: [Asterisk-Users] Newbie: Crossing my fingers
That really help me: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+files miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
I just got back from Boston where we completed testing of the TE410P for FCC, Euro, and Australian approvals, and I'm happy to say we passed all our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as surges) for both telco and leased line applications. Hopefully we'll have the official documents soon, but I know there are a lot of you out there that are happy to hear that. Mark p.s. We were the *first* independent PRI implementation to come through that lab! Of all the units they've tested, we're the first to choose the build path on build vs. buy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi newbie questions
Hi I have to PCI AVM cards on asterisk and i want to configure them with chan_capi. But there are some things that need explanation to me. 1. If i have not declared MSNs to my provider am i going to put msn=0 to capi.conf? 2.How do the 'deflate' and 'incomingmsn' parameters be set? 3. Although i have two controllers and i4l load them both, chan_capi installs only one? Is there something wrong with capi.conf? Thank you __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
Welcome I have been updating this doc with links to user documenation as i come across it http://bugs.digium.com/bug_view_page.php?bug_id=070 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
Excellent news, congratulations !! -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 15:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P I just got back from Boston where we completed testing of the TE410P for FCC, Euro, and Australian approvals, and I'm happy to say we passed all our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as surges) for both telco and leased line applications. Hopefully we'll have the official documents soon, but I know there are a lot of you out there that are happy to hear that. Mark p.s. We were the *first* independent PRI implementation to come through that lab! Of all the units they've tested, we're the first to choose the build path on build vs. buy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX calling number
On Fri, 2003-09-26 at 04:09, Chee Foong wrote:
Ahh...Understood. That's possible.
But my problem is I will have 500 users (and increasing). I can't have an
entry for every users in the config file. The only way to handle this so far
I found is to use number as username, therefore we can use only 1 extension:
exten = _700XX,1,Dial(IAX/${EXTEN})
But user wont like it if username is a long string of number, they prefer
meaningful name.
You get it one way or the other. Either the username is a number you can
map, or 1 more line of extensions.conf per user. Your going to put that
same effort into the iax.conf file anyways so don't complain about where
it is.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Newbie: Crossing my fingers
Any one else getting access denied to this bug? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Friday, 26 September 2003 11:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie: Crossing my fingers Welcome I have been updating this doc with links to user documenation as i come across it http://bugs.digium.com/bug_view_page.php?bug_id=070 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trouble with MGCP Phone
Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line = aaln/2 ;line = aaln/1 [192.168.10.10] host = 192.168.10.10 context = default line = aaln/1 I haven't found any mgcp related information Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
On Fri, Sep 26, 2003 at 06:40:12AM -0700, TC wrote: Welcome I have been updating this doc with links to user documenation as i come across it http://bugs.digium.com/bug_view_page.php?bug_id=070 ERROR: Access Denied. as user anonymous... guess I need to create an account. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
- Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 11:37 PM Subject: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P I just got back from Boston where we completed testing of the TE410P for FCC, Euro, and Australian approvals, and I'm happy to say we passed all our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as surges) for both telco and leased line applications. Hopefully we'll have the official documents soon, but I know there are a lot of you out there that are happy to hear that. Congratulations Mark, and everyone else involved. I don't know about the other approvals, but the ACA approvals for Australia are apparently really difficult to get (assuming this is what you meant by Australian approvals) - so congratulations again. Mark p.s. We were the *first* independent PRI implementation to come through that lab! Of all the units they've tested, we're the first to choose the build path on build vs. buy. Excellent! :-) -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] difference between nufone chan_h323 and openh323 chan_oh323
Title: Message can anyone tell me the real difference? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au
[Asterisk-Users] Wildcard for Conferencing (VoIP)
Hello: Our system only uses VoIP (OH323), and we would like to incorporate Conferencing (Meetme) and Music on Hold. We would like to order a Digium Wildcard (not for real telephony use, only to efficiently support Conferencing and MoH). Is X100P the appropriate card? Does E100P offer more performance for this purpose? Thank you, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATM support?
Is there any interest in having ATM support for the various digium T1 cards? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '.' pattern and non-SIP phones
*8 is call pickup. Can you choose a different extension? If not, I'm going to have to make call pickup not be checked if you don't have a pickup/ring group. As I explained on IRC, I do not think this is interfering. I am currently using *1, *2, *3, *6, and *7 without issue. If I move this to *1 (which jumps to the demo right now), I have the same problem. I move it to plain old 8 and it has the same issue. Offhand, can these builtin * codes be eliminated? I am using straight dialling (no 9 or any other prefix for a normal call) and would like to use *-codes for the special stuff. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P - Busydetect / calls being disconnected - Australia; tip.
Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still happened. I thought that there was some remote possibility that the phone company was reversing the line on answering long distance calls, so I switched to fxs_ls instead of fxs_ks - no difference. Various things were tried to no avail, until I made a long distance call over a different carrier to our usual carrier (we use Optus, I made the call over Telstra). When the remote end answered, my end disconnected. What was happening was, when the call is answered, 5 quick chirps are sent down the line. However, because of the bug in the Cisco 7960 causing the first 1/2 a second or so of a conversation to be cut off - I didn't hear these chirps and as such I didn't think of the next bit: Basically, because I had busycount set to 3 and busydetect set to yes, these chirps were being detected by the busydetect function and causing the call to be disconnected. I raised the busycount to something safe (8) and this no longer happened. This has me worried for a while, especially as I'd just disconnected the old PBX a few days ago and spent a nice amount of money on Cisco 7960 and 7940 IP phones (and will probably be ordering more in the near future). Anyway, I'm pleased to report that everything is now working perfectly and I'm extremely happy with Asterisk. I'd contribute, but alas I'm not much of a C programmer. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to use VoiceMail2 mysql table structure?
Hi list, I would be interested in knowing in how to use voicemail2 stored in database to use in extensions.conf ie what changes in which conf file will enable the use of this database. Rgds Manoj K Gupta CREATE TABLE users ( context char(79) DEFAULT '' NOT NULL, mailbox char(79) DEFAULT '' NOT NULL, password char(79) DEFAULT '' NOT NULL, fullname char(79) DEFAULT '' NOT NULL, email char(79) DEFAULT '' NOT NULL, pager char(79) DEFAULT '' NOT NULL, options char(159) DEFAULT '' NOT NULL, stamp timestamp, PRIMARY KEY (context,mailbox) ); ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] initial review of Grandstream HT-286 ATA device
The PDF on the website says that this thing supports a downloadable ring-tone. This makes me somewhat suspicious - does this thing generate ringing voltages and actually ring the attached analog phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. In the exerpt from the config you posted (below), the destination in the dial-peer is the same as the address on the enternet interface. Perhaps this is not your actual config? In any event, if this is correct, it should work. dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
Hi, I've asked them about the switch and they told me that its a Siemens EWSD.. Regards Oz On 9/25/03 11:32 AM, Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: Hi. Do you know what switch your telco has ? The one they are using to provide you the service. Osvaldo Mundim Junior wrote: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
Some ISO firmwares have bugs.. we ran into this and had to downgrade to get it to work correctly. bkw On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have fixed i already. And still it does not want to work :( - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 11:58 AM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. In the exerpt from the config you posted (below), the destination in the dial-peer is the same as the address on the enternet interface. Perhaps this is not your actual config? In any event, if this is correct, it should work. dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMailMain skipping extension and password prompting
OK, Here is a down and dirty which will work in limited situations (like
when there are not to many extensions to re-define - which is one of the
things I want to avoid)... The channel is the first parameter passed to
[globals]
Zap/5-=s6147
Zap/16=s6158
exten = 6199,1,GoToIf(${${CHANNEL:0:6}}?6199|2:6199|4)
exten = 6199,2,VoicemailMain2(${${CHANNEL:0:6}})
exten = 6199,3,Hangup
exten = 6199,4,VoicemailMain2
exten = 6199,5,Hangup
John
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Re: [Asterisk-Users] Cisco 2600 and ASTERISK
I made it work! My miste was: session target ipv4:66.178.36.220:1720 when I change it to session target ipv4:66.178.36.220 everything works just fine. Right now I have to make outgoing call. - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 12:34 PM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK Some ISO firmwares have bugs.. we ran into this and had to downgrade to get it to work correctly. bkw On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have fixed i already. And still it does not want to work :( - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 11:58 AM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. In the exerpt from the config you posted (below), the destination in the dial-peer is the same as the address on the enternet interface. Perhaps this is not your actual config? In any event, if this is correct, it should work. dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
Ha, looks like it got marked as private view, can someone from DigiumAdmin make it Public Viewable -Original Message- From: PJ Welsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: September 26, 2003 7:27 AM Subject: Re: [Asterisk-Users] Newbie: Crossing my fingers On Fri, Sep 26, 2003 at 06:40:12AM -0700, TC wrote: Welcome I have been updating this doc with links to user documenation as i come across it http://bugs.digium.com/bug_view_page.php?bug_id=070 ERROR: Access Denied. as user anonymous... guess I need to create an account. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP codecs Errors
Me too. Please fix this soon please somebody. MATT--- -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Friday, September 26, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP codecs Errors I think there is a problem in CVS because yesterday I updated Asterisk from CVS and I had the same problem with codecs. When I went back with CVS everything was working again normal. -- Bart - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 8:22 PM Subject: Re: [Asterisk-Users] SIP codecs Errors On Thursday 25 September 2003 15:01, CW_ASN wrote: Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The show codecs command shows: *CLI show codecs 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video 'show codecs' in no way, shape, or form indicates what codecs are useable in your sip config. It returns the same output on ALL machines. Look in your sip.conf for what codecs you have available. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite for Linux
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 No, not really. But X-Lite for Windows works just fine running under Wine on Linux. You have to change some of the auto detection features, specifically auto detection of the IP address. You must manually enter the IP address that X-Lite will use, and then it works perfectly. Now I have a decent cross-platform SIP softphone client. :) (now if only my laptop could record sound, I could really use it) - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGTeuYsUrHkpYtARAjZcAJ9wyvfuaP8bcV1OpcYcOG9CXbgdlgCfe6Py h5TQZgIFlT21dxbNfR6zrT8= =Isbq -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Mailscanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
No, not really. But X-Lite for Windows works just fine running under Wine on Linux. I thought gnophone and kphone both worked well? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
Small problem with that theory, It look slike Xten have removed Transfer functionality from the X-Lite product.. Which then poses the question.. Buy the X-Pro product for $50 and a fair quality headset for $20 (total $70) or buy a hardphone for $65 and be able to run it independently of the PC.. Hmm.. Tuff choice!! Later.. Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 No, not really. But X-Lite for Windows works just fine running under Wine on Linux. You have to change some of the auto detection features, specifically auto detection of the IP address. You must manually enter the IP address that X-Lite will use, and then it works perfectly. Now I have a decent cross-platform SIP softphone client. :) (now if only my laptop could record sound, I could really use it) - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGTeuYsUrHkpYtARAjZcAJ9wyvfuaP8bcV1OpcYcOG9CXbgdlgCfe6Py h5TQZgIFlT21dxbNfR6zrT8= =Isbq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Config TE410P + TDM400
When configuring a TE410P which is only attached to a single E1 together with a TDM400, how should one count the channels for the next Zap interface? Must I put 4 span lines in zapata.conf and define all channels up to 124? thus having the TDM400's start at 125? Or can I comment out the 3 spans I don't use and start at channel 32 for the TDM400? (this would get nasty when adding extra lines, but would stop asterisk from trying to look at E1's which are not connected) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Kohlsmith wrote: |No, not really. But X-Lite for Windows works just fine running under |Wine on Linux. | | | I thought gnophone and kphone both worked well? They do, for what they do. gnophone is (to my knowledge) and IAX-only client. kphone (please someone enlighten me) cannot inject DTMF digits into the stream, at least I have not found a digit pad or similar way to do it. I was finally able to get linphone to work with * by having * trust the IP address linphone was coming from. linphone apparently cannot correctly authenticate to * properly. Don't know what's going on there. linphone does have the ability to inject DTMF digits and seems to work properly from that respect. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGn3uYsUrHkpYtARAu8lAJ9Ts6It2BCGDzh0uwTqsXbHeC1QBgCaA8et AbKRrGevO84pgthNB03D7wU= =+cKf -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Mailscanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
really... it does transfers, but only with the Bye/Also transfer method (not supported by *, I think). so is the same as not having it ;) matteo. Il ven, 2003-09-26 alle 18:25, WipeOut ha scritto: Small problem with that theory, It look slike Xten have removed Transfer functionality from the X-Lite product.. Which then poses the question.. Buy the X-Pro product for $50 and a fair quality headset for $20 (total $70) or buy a hardphone for $65 and be able to run it independently of the PC.. Hmm.. Tuff choice!! Later.. Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 No, not really. But X-Lite for Windows works just fine running under Wine on Linux. You have to change some of the auto detection features, specifically auto detection of the IP address. You must manually enter the IP address that X-Lite will use, and then it works perfectly. Now I have a decent cross-platform SIP softphone client. :) (now if only my laptop could record sound, I could really use it) - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGTeuYsUrHkpYtARAjZcAJ9wyvfuaP8bcV1OpcYcOG9CXbgdlgCfe6Py h5TQZgIFlT21dxbNfR6zrT8= =Isbq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 WipeOut wrote: | Small problem with that theory, It look slike Xten have removed | Transfer functionality from the X-Lite product.. And you only get 3 lines, where with the X-Pro version you get 6. | Which then poses the question.. | | Buy the X-Pro product for $50 and a fair quality headset for $20 (total | $70) or buy a hardphone for $65 and be able to run it independently of | the PC.. | | Hmm.. Tuff choice!! Good point. Although, the more X-Pro licenses you purchase, the cheaper they get. Don't know if that's the same for hardphones. But, why buy a headset for the softphone? You can do echo cancellation, right? - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGp4uYsUrHkpYtARAu3OAJ98lDTlK30j7OZ9Rk4asC3YRHLkHwCdGVoc j4t/0E4Im+FTYk4ww9mQF1g= =0JUj -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Mailscanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P - Busydetect / calls being disconnected - Australia; tip.
Because of the nature of busydetect algorithm busycount shouldn't be set to less than 8. It's 10 by default. Just imagine that you dial a number that is attached to some speed dial key. It'll surely cause hangup if busydetect 8. Martin On Sat, 27 Sep 2003, Shaun Ewing wrote: Hi All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still happened. I thought that there was some remote possibility that the phone company was reversing the line on answering long distance calls, so I switched to fxs_ls instead of fxs_ks - no difference. Various things were tried to no avail, until I made a long distance call over a different carrier to our usual carrier (we use Optus, I made the call over Telstra). When the remote end answered, my end disconnected. What was happening was, when the call is answered, 5 quick chirps are sent down the line. However, because of the bug in the Cisco 7960 causing the first 1/2 a second or so of a conversation to be cut off - I didn't hear these chirps and as such I didn't think of the next bit: Basically, because I had busycount set to 3 and busydetect set to yes, these chirps were being detected by the busydetect function and causing the call to be disconnected. I raised the busycount to something safe (8) and this no longer happened. This has me worried for a while, especially as I'd just disconnected the old PBX a few days ago and spent a nice amount of money on Cisco 7960 and 7940 IP phones (and will probably be ordering more in the near future). Anyway, I'm pleased to report that everything is now working perfectly and I'm extremely happy with Asterisk. I'd contribute, but alas I'm not much of a C programmer. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATM support?
On Fri, 26 Sep 2003, Ryan Butler wrote: On Fri, 2003-09-26 at 09:16, Dave Weis wrote: Is there any interest in having ATM support for the various digium T1 cards? If you mean ATM as well as IMA muxing of ATM T1's (IMA 1.1 please), and the ability for having atm interfaces and pri's on a quad t1 card, then yes :) Do you use pvc's or svc's for voice? Where can I get the correct docs to do this? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite for Linux
I have just installed the latest version downloaded this afternoon (about 1 hour ago) and the transfer button look to be completely dissabled.. in otherwords you click it but nothing happen's. :( Oh well I guess it was nice while it lasted.. :) Brancaleoni Matteo wrote: really... it does transfers, but only with the Bye/Also transfer method (not supported by *, I think). so is the same as not having it ;) matteo. Il ven, 2003-09-26 alle 18:25, WipeOut ha scritto: Small problem with that theory, It look slike Xten have removed Transfer functionality from the X-Lite product.. Which then poses the question.. Buy the X-Pro product for $50 and a fair quality headset for $20 (total $70) or buy a hardphone for $65 and be able to run it independently of the PC.. Hmm.. Tuff choice!! Later.. Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 No, not really. But X-Lite for Windows works just fine running under Wine on Linux. You have to change some of the auto detection features, specifically auto detection of the IP address. You must manually enter the IP address that X-Lite will use, and then it works perfectly. Now I have a decent cross-platform SIP softphone client. :) (now if only my laptop could record sound, I could really use it) - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/dGTeuYsUrHkpYtARAjZcAJ9wyvfuaP8bcV1OpcYcOG9CXbgdlgCfe6Py h5TQZgIFlT21dxbNfR6zrT8= =Isbq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
On Fri, 26 Sep 2003 12:51:16 -0300, Bartosz Jozwiak wrote I made it work! My miste was: session target ipv4:66.178.36.220:1720 when I change it to session target ipv4:66.178.36.220 everything works just fine. Do you get the called number? So you can decide what to do woth the call. I'm really interestet in this because i'm stuck at this point. I'll get the calls but i don't get the called number. Right now I have to make outgoing call. - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 12:34 PM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK Some ISO firmwares have bugs.. we ran into this and had to downgrade to get it to work correctly. bkw On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have fixed i already. And still it does not want to work :( - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 11:58 AM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. In the exerpt from the config you posted (below), the destination in the dial-peer is the same as the address on the enternet interface. Perhaps this is not your actual config? In any event, if this is correct, it should work. dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Visit anne2d.tznetz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P - Busydetect / calls being disconnected - Australia; tip.
- Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 27, 2003 2:36 AM Subject: Re: [Asterisk-Users] X100P - Busydetect / calls being disconnected - Australia; tip. Because of the nature of busydetect algorithm busycount shouldn't be set to less than 8. It's 10 by default. Just imagine that you dial a number that is attached to some speed dial key. It'll surely cause hangup if busydetect 8. Martin Aah, thanks for the tip Martin; that I wasn't aware of. Maybe I'll try 10 :-) -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI: getting the return code from an exec()'d application?
Steven J. Sobol wrote: On Fri, 26 Sep 2003, Chee Foong wrote: Hello Steven, I am planing to do the same thing: make dial return correct dial status and use agi to detect it. Is it possible for you to share the modified dial source Oh, sure, getting ready to go to bed but can do it tomorrow. A diff would probably be the best thing to do, unless people have a problem with me posting the source as an attachment? If i am not mistaken, result return by exec is like: 200 Result=number additional information, if any Yeah, but I'm using James's AGI perl module here. He tells me that the exec() return code SHOULD be the return code of the application that gets executed. That, unfortunately, isn't happening. Update - James e-mailed me and asked for a copy of app_agidial.c. Said he wants to see if he can get the problem resolved before he leaves for the weekend. I gave it to him, but if he can't resolve the issue, I'll post the source here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE:RE: [Asterisk-Users] G729 experiences.. (fwd)
Ww! Kick this guy out from this list and pls filter [EMAIL PROTECTED] It's very annoying Isamar -- Forwarded message -- Date: Fri, 26 Sep 2003 08:48:45 -0300 (BRT) From: AntiSpam UOL [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE:RE: [Asterisk-Users] G729 experiences.. [antispam_txt.gif] Olá, Você enviou uma mensagem para [EMAIL PROTECTED] Para que sua mensagem seja encaminhada, por favor, clique aqui Esta confirmação é necessária porque [EMAIL PROTECTED] usa o Antispam UOL, um programa que elimina mensagens enviadas por robôs, como pornografia, propaganda e correntes. As próximas mensagens enviadas para [EMAIL PROTECTED] não precisarão ser confirmadas*. *Caso você receba outro pedido de confirmação, por favor, peça para [EMAIL PROTECTED] incluí-lo em sua lista de autorizados. Atenção! Se você não conseguir clicar no atalho acima, acesse este endereço: http://tira-teima.as.uol.com.br/challengeSender.html?data=ETGT78UpKvC hfzyhxiozM34znL%2FHAxIUxbtjYBksCfIkh9Vpz6uXkDLENcDcbtWR%2FzId%2B5CarS %0ApSXSoIoUZj4mmHOWjdq6vGioCSDHpLUVsqsRF%2FBEGuNMcL8uYQPJbUIVE2CP%2F0 IZP1oJs3xJc7v%0AaACy540CC4fohViLaUc%3D __ Hi, You´ve just sent a message to [EMAIL PROTECTED] In order to confirm the sent message, please click here This confirmation is necessary because [EMAIL PROTECTED] uses Antispam UOL, a service that avoids unwanted messages like advertising, pornography, viruses, and spams. Other messages sent to [EMAIL PROTECTED] won't need to be confirmed*. *If you receive another confirmation request, please ask [EMAIL PROTECTED] to include you in his/her authorized e-mail list. Warning! If the link doesn´t work, please copy the address below and paste it on your browser: http://tira-teima.as.uol.com.br/challengeSender.html?data=ETGT78UpKvC hfzyhxiozM34znL%2FHAxIUxbtjYBksCfIkh9Vpz6uXkDLENcDcbtWR%2FzId%2B5CarS %0ApSXSoIoUZj4mmHOWjdq6vGioCSDHpLUVsqsRF%2FBEGuNMcL8uYQPJbUIVE2CP%2F0 IZP1oJs3xJc7v%0AaACy540CC4fohViLaUc%3D Use o AntiSpam UOL e proteja sua caixa postal RdFhCfCggWeb = '1996-'; ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
To get callerid you apparently need the VIC-2FXO-M1 card from Cisco. Gerry At 12:07 PM 9/26/2003, Michael Gschwandtner wrote: On Fri, 26 Sep 2003 12:51:16 -0300, Bartosz Jozwiak wrote I made it work! My miste was: session target ipv4:66.178.36.220:1720 when I change it to session target ipv4:66.178.36.220 everything works just fine. Do you get the called number? So you can decide what to do woth the call. I'm really interestet in this because i'm stuck at this point. I'll get the calls but i don't get the called number. Right now I have to make outgoing call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P: Can I detect/react to CLASS you got voicemail signals?
The subject says it all... I have an X100P and I have (for now anyway) Bell Canada's Call Answer which will notify you through one of those nifty CLASS signals that you either do or do not have voicemail. This is not only a stuttering dialtone but some actual signal passed so CLASS-aware phones can detect it and flash their message waiting indicator. I'd like to detect this (and any other of these type of signal) in asterisk, but am not sure how to do so. Is it possible? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing out with the outgoing queue problem.
Hi,
I have cvs updated
all my modules (zapata, libpri, zaptel andasterisk).
I have also read in
the archives seems that no-one has run into this
problem.
What I'm trying to
do is simple. Just make and outbound call using the
/var/spool/asterisk/outgoing
directory.
I copied
/usr/src/asterisk/sample.call and only changed the context
extension.
I
configured my Zap1 to the same context. I havetwo X100P (Zap1
Zap2)and
oneS100U (Zap3).
If I
use my S100U and dial extension 800, it works. It
calls.
However
when I copy my 1.call file. it says:
Unable
to create channel of type 'Zap'.
Does
anyone have any suggestions? or know what am I
missing?
Thanks,
Dante
Here's
my configuration:
extensions:
[callme]
...
exten =
800,1,Dial(Zap/1/19548738986)exten =
800,2,BackGround(demo-congrats)exten =
800,3,BackGround(demo-instruct)
...
1.call:
Channel:
Zap/1MaxRetries: 2RetryTime: 60WaitTime: 30Context:
callmeExtension: 800
...
cp
/tmp/1.call /var/spool/asterisk/outgoing
-- Attempting call on Zap/1 for [EMAIL PROTECTED]:1 (Retry 1) --
Executing Dial("Zap/1-1", "Zap/1/195487389xx") in new stackNOTICE[360464]:
File app_dial.c, Line 499 (dial_exec): Unable to
create channel of type 'Zap' == Everyone is busy at
this time -- Executing BackGround("Zap/1-1",
"demo-congrats") in new stack -- Playing
'demo-congrats'
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
That's exactly what I encountered every timeI eventually gave up on it...Brian (bkw) gave me the configs and all, but was unable to get it to work -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Friday, September 26, 2003 11:24 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK I just get all time fast bussy signal. - Original Message - From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 11:58 AM Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK On Fri, 26 Sep 2003, Bartosz Jozwiak wrote: I have pointed it to Asterisk for sure not to local cisco ethernet. I think there is something wrong with the router. In the exerpt from the config you posted (below), the destination in the dial-peer is the same as the address on the enternet interface. Perhaps this is not your actual config? In any event, if this is correct, it should work. dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK and calling out
Like Gerry wrote for callerid you need VIC-2FXO-M1 card.
Right now I am stuck on making outgoing call.
Could soembody help me with the configuration.
On cisco I have soemthing like that:
dial-peer voice 400 pots
destination-pattern 9T
voice-port 1/1/0
In extensions.conf
exten=9.,1,Dial(H323/${EXTEN:[EMAIL PROTECTED])
exten=9.,2,Congestion
In h323.conf
[blah]
type=friend
host=xxx.xxx.xxx.xxx
contex=default
exten=,1,Goto(s,1)
incominglimit=4
When dialing
99xx
I am getting 503 Service Unavailable
And there is nothing on debug on Router. Looks like call get stuck somewhere
in asterisk
And Asterisk says this:
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
-- Executing Congestion(SIP/1008-1dc7, ) in new stack
== Spawn extension (default, 9908500569, 2) exited non-zero on
'SIP/1008-1dc7'
What could be the problem ???
- Original Message -
From: Gerry Boudreaux [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 26, 2003 2:26 PM
Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK
To get callerid you apparently need the VIC-2FXO-M1 card from Cisco.
Gerry
At 12:07 PM 9/26/2003, Michael Gschwandtner wrote:
On Fri, 26 Sep 2003 12:51:16 -0300, Bartosz Jozwiak wrote
I made it work!
My miste was:
session target ipv4:66.178.36.220:1720
when I change it to session target ipv4:66.178.36.220
everything works just fine.
Do you get the called number? So you can decide what to do woth the
call. I'm really
interestet in this because i'm stuck at this point. I'll get the calls
but
i don't get the called
number.
Right now I have to make outgoing call.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing out with the outgoing queue problem.
On Fri, 2003-09-26 at 12:35, Dante Alzamora wrote: extensions: [callme] ... exten = 800,1,Dial(Zap/1/19548738986) exten = 800,2,BackGround(demo-congrats) exten = 800,3,BackGround(demo-instruct) ... 1.call: Channel: Zap/1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 ... cp /tmp/1.call /var/spool/asterisk/outgoing -- Attempting call on Zap/1 for [EMAIL PROTECTED]:1 (Retry 1) -- Executing Dial(Zap/1-1, Zap/1/195487389xx) in new stack NOTICE[360464]: File app_dial.c, Line 499 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing BackGround(Zap/1-1, demo-congrats) in new stack -- Playing 'demo-congrats' This is is pretty straight forward, you are using Zap/1 to pickup and your dial command is also trying to use Zap/1. You have it busy in one part of the call and the second part can't do it's job. BTW, you should not have a extension layout that is a dial, then background commands. The normal end of a dial command is a hangup or busy. What you probably want is something along the ways of this for your 1.call Channel: Zap/1/19548738986 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 This will make the outbound call and the play your messages. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing out with the outgoing queue problem.
That did the trick. Thanks Steven, Dante -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Friday, September 26, 2003 2:04 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialing out with the outgoing queue problem. On Fri, 2003-09-26 at 12:35, Dante Alzamora wrote: extensions: [callme] ... exten = 800,1,Dial(Zap/1/19548738986) exten = 800,2,BackGround(demo-congrats) exten = 800,3,BackGround(demo-instruct) ... 1.call: Channel: Zap/1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 ... cp /tmp/1.call /var/spool/asterisk/outgoing -- Attempting call on Zap/1 for [EMAIL PROTECTED]:1 (Retry 1) -- Executing Dial(Zap/1-1, Zap/1/195487389xx) in new stack NOTICE[360464]: File app_dial.c, Line 499 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing BackGround(Zap/1-1, demo-congrats) in new stack -- Playing 'demo-congrats' This is is pretty straight forward, you are using Zap/1 to pickup and your dial command is also trying to use Zap/1. You have it busy in one part of the call and the second part can't do it's job. BTW, you should not have a extension layout that is a dial, then background commands. The normal end of a dial command is a hangup or busy. What you probably want is something along the ways of this for your 1.call Channel: Zap/1/19548738986 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: callme Extension: 800 Priority: 2 This will make the outbound call and the play your messages. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman and SIP?
I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = SIP\3846) Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The Green light does light up. What should Invite and Originate do, right now they just ring a phone once and hangup. Anyone know of any other programs that I can be tested for call status and redirection? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gastman and SIP?
On Friday 26 September 2003 02:03 pm, James Sizemore wrote: I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = SIP\3846) Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The Green light does light up. That's also how Zap channels work. Unfortunately, SIP channels are a little more difficult to link to a particular icon, as SIP channels are created and destroyed on the fly and channel numbers are not reused (note that SIP/3846 means that this is the 3,846th SIP session since asterisk was last restarted). What should Invite and Originate do, right now they just ring a phone once and hangup. Anyone know of any other programs that I can be tested for call status and redirection? Originate should allow a call to be started from the GUI. The originating channel rings, then the call is started as if the extension entered in the GUI was actually dialled on that channel. I'm not sure what invite should do, though. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incomming call management
Hi all, I'm looking for the following functionality: if my queues reach a certain threshold, I would like to disable any available zap / PRI channels, so my telco doesn't try to connect more people. After a while, I will enable them again. Any hints on how to implement this? Should I be looking to patch * on chan_zap level, or should I somehow ioctl zapata and disable these channels somehow? Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to find a path from ULAW to G723
Hello, I just CVS'd today and now I'm getting these errors when I call one grandstream phone to another both using 711U: NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW The calls connect just fine and I'm not using 723 ANYWHERE, where is this NOTICE coming from? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard for Conferencing (VoIP)
We would like to order a Digium Wildcard (not for real telephony use, only to efficiently support Conferencing and MoH). Is X100P the appropriate card? Does E100P offer more performance for this purpose? X100P is fine. There would be no noticible benefit from the E100P unless you plan to use it with E1. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the g729 situation
Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server You can call us for free support on G.729 if you purchased it from us. 877-LINUX-ME just choose install support. Now I wrote to vonage as per the instructions further in the error message, requesting a certificate. I'm sure I'm not the only one going through all these hoops. I trust you mean Voiceage not Vonage but in any case neither will likely be useful. Definitely should contact us directly. - is there a cracked g729 binary out there? (which I plan to use inside my license agreement) Not as far as I know. - is it true that * has to be run with -c when using g729 ? Yes, again we're trying to get Voiceage to fix the issue, but working with closed source, slow moving, intellectual property based vendors is generally a pretty miserable experience. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK and calling out
You have no dial-peer telling the router what to do with the outbound call.
http://www.tape.net/~gerry/asterisk/cisco26x0.html
At 12:50 PM 9/26/2003, you wrote:
Like Gerry wrote for callerid you need VIC-2FXO-M1 card.
Right now I am stuck on making outgoing call.
Could soembody help me with the configuration.
On cisco I have soemthing like that:
dial-peer voice 400 pots
destination-pattern 9T
voice-port 1/1/0
In extensions.conf
exten=9.,1,Dial(H323/${EXTEN:[EMAIL PROTECTED])
exten=9.,2,Congestion
In h323.conf
[blah]
type=friend
host=xxx.xxx.xxx.xxx
contex=default
exten=,1,Goto(s,1)
incominglimit=4
When dialing
99xx
I am getting 503 Service Unavailable
And there is nothing on debug on Router. Looks like call get stuck somewhere
in asterisk
And Asterisk says this:
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
-- Executing Congestion(SIP/1008-1dc7, ) in new stack
== Spawn extension (default, 9908500569, 2) exited non-zero on
'SIP/1008-1dc7'
What could be the problem ???
- Original Message -
From: Gerry Boudreaux [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 26, 2003 2:26 PM
Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK
To get callerid you apparently need the VIC-2FXO-M1 card from Cisco.
Gerry
At 12:07 PM 9/26/2003, Michael Gschwandtner wrote:
On Fri, 26 Sep 2003 12:51:16 -0300, Bartosz Jozwiak wrote
I made it work!
My miste was:
session target ipv4:66.178.36.220:1720
when I change it to session target ipv4:66.178.36.220
everything works just fine.
Do you get the called number? So you can decide what to do woth the
call. I'm really
interestet in this because i'm stuck at this point. I'll get the calls
but
i don't get the called
number.
Right now I have to make outgoing call.
___
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[Asterisk-Users] number detection problem.
Hello, We're using asterisk with a z-plex 10 and a Zap tormentia card at a non-profit home for the aged. Sometimes when we dial out, our numbers are misintepreted with one number being detected twice. For instance 337-4411 becomes 337-7441 7's 4's and 0's seem to be particularly prone to this issue. Any ideas? I've tried turning down rxgain and txgain. A little debugging info: # uname -a Linux pbx 2.4.18 #1 Mon Aug 25 14:27:34 EDT 2003 i686 unknown Debian system 1.2Ghz Athalon Thunderbird Asterisk CVS-08/25/03-14:35:09, Copyright (C) 1999-2001 Linux Support Services, Inc. Thanks, Jon
Re: [Asterisk-Users] the g729 situation
Hi, On the same note, I am having a problem with G.729, having 4 * asterisk boxes 2 with 10 licenses and one with 2 licenses. The licensesinstalls fine, but the codec doesn't work as supposed to be. In path thru situation, where a UA (grandstream phone) is talking to the * that is connected to NuFone over IAX/2 seems to work. But when NuFone stopped supporting G729. The RTP path could not be established (G729-*-SPEEX). However, the following scenario works (G711/GSM-*-SPEEX) Thanks for your helpMark Spencer [EMAIL PROTECTED] wrote: Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our serverYou can call us for free support on G.729 if you purchased it from us.877-LINUX-ME just choose "install support". Now I wrote to vonage as per the instructions further in the error message, requesting a certificate. I'm sure I'm not the only one going through all these hoops.I trust you mean "Voiceage" not "Vonage" but in any case neither willlikely be useful. Definitely should contact us directly. - is there a cracked g729 b inary out there? (which I plan to use inside my license agreement)Not as far as I know. - is it true that * has to be run with -c when using g729 ?Yes, again we're trying to get Voiceage to fix the issue, but working withclosed source, slow moving, intellectual property based vendors isgenerally a pretty miserable experience.Mark___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] the g729 situation
NuFone only had 3 G.729 licenses and when we went to add more it blew up our system and now we have none. Anyways, we are not very fond of the VoiceAge licensing terms. We prefer iLBC. Jeremy McNamara Thomas Moghnie wrote: Hi, On the same note, I am having a problem with G.729, having 4 * asterisk boxes 2 with 10 licenses and one with 2 licenses. The licenses installs fine, but the codec doesn't work as supposed to be. In path thru situation, where a UA (grandstream phone) is talking to the * that is connected to NuFone over IAX/2 seems to work. But when NuFone stopped supporting G729. The RTP path could not be established (G729-*-SPEEX). However, the following scenario works (G711/GSM-*-SPEEX) Thanks for your help */Mark Spencer [EMAIL PROTECTED]/* wrote: Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server You can call us for free support on G.729 if you purchased it from us. 877-LINUX-ME just choose install support. Now I wrote to vonage as per the instructions further in the error message, requesting a certificate. I'm sure I'm not the only one going through all these hoops. I trust you mean Voiceage not Vonage but in any case neither will likely be useful. Definitely should contact us directly. - is there a cracked g729 b inary out there? (which I plan to use inside my license agreement) Not as far as I know. - is it true that * has to be run with -c when using g729 ? Yes, again we're trying to get Voiceage to fix the issue, but working with closed source, slow moving, intellectual property based vendors is generally a pretty miserable experience. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping http://shopping.yahoo.com/?__yltc=s%3A15443%2Cd%3A22708228%2Cslk%3Atext%2Csec%3Amail - with improved product search ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choppy communication issue
I've setup a trial Asterisk install based on the RH8 install guide mentioned on this list. (Thanks, Andy!) I've configured two other working systems with SJPhone software for SIP. However, while I can call the phones, the communication is choppy. About every three or four seconds it cuts out briefly and then returns to normal. Perhaps someone can give me some pointers. My setups are described below: 1. Asterisk server 2.6 Athlon AMD / 512 MB Ram Using separate network card. Using built in sound card. (This should only cause problems for voice mail and prompts, correct?) 2. Client machine 1 1Ghz Celeron, 256mb Ram, etc. 3. Client machine 2 500 Mhz AMD, 96 mb Ram, etc. They are connected via a BayNetworks 350F-HD switch. (Fully capable, however, it does not have advanced packet routing/ prioritizing.) Is it practical to assume that the problem is in the switch? (ie. needing QoS for VOIP packets) Or perhaps I should be looking somewhere else to resolve this. I would appreciate any pointers or suggestions offered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
Congratulations to the team at Digium! That is great progress. Peter At 08:37 26/09/2003 -0500, you wrote: I just got back from Boston where we completed testing of the TE410P for FCC, Euro, and Australian approvals, and I'm happy to say we passed all our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as surges) for both telco and leased line applications. Hopefully we'll have the official documents soon, but I know there are a lot of you out there that are happy to hear that. Mark p.s. We were the *first* independent PRI implementation to come through that lab! Of all the units they've tested, we're the first to choose the build path on build vs. buy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the g729 situation
I totally agree with you. The codec is buggy and the license agreement from VoiceAge is - to put it in proper way- preposterous. However, I have to find a solution for customers with Cisco (79xx and budgetone) that don't want to use up all their network bandwidth. Until someone implements speex of iLBC on those phone, we're stuck with G.729 RegardsJeremy McNamara [EMAIL PROTECTED] wrote: NuFone only had 3 G.729 licenses and when we went to add more it blew up our system and now we have none. Anyways, we are not very fond of the VoiceAge licensing terms. We prefer iLBC.Jeremy McNamaraThomas Moghnie wrote: Hi, On the same note, I am having a problem with G.729, having 4 * asterisk boxes 2 with 10 licenses and one with 2 licenses. The licenses installs fine, but the codec doesn't work as supposed to be. In path thru situation, where a UA (grandstream phone) is talking to the * that is connected to NuFone over IAX/2 seems to work. But when NuFone stopped supporting G729. The RTP path could not be established (G729-*-SPEEX). However, the following scenario works (G711/GSM-*-SPEEX) Thanks for your help */Mark Spencer <[EMAIL PROTECTED]>/* wrote: Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server You can call us for free support on G.729 if you purchased it from us. 877-LINUX-ME just choose "install support". Now I wrote to vonage as per the instructions further in the error message, requesting a certificate. I'm sure I'm not the only one going through all these hoops. I trust you mean "Voiceage" not "Vonage" but in any case neither will likely be useful. Definitely should contact us directly. - is there a cracked g729 b inary out there? (which I plan to use inside my license agreement) Not as far as I know. - is it true that * has to be run with -c when using g729 ? Yes, again we're trying to get Voiceage to fix the issue, but working with closed source, slow moving, intellectual property based vendors is generally a pretty miserable experience. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping - with improved product search ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] initial review of Grandstream HT-286 ATA device
I'm not sure about ring tones It does produce the 90 VRMS ring signal and cause the analog phone to ring. On Fri, Sep 26, 2003 at 09:55:45AM -0500, Peter Pauly wrote: The PDF on the website says that this thing supports a downloadable ring-tone. This makes me somewhat suspicious - does this thing generate ringing voltages and actually ring the attached analog phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] number detection problem.
On Fri, 2003-09-26 at 15:09, Jon Hopper wrote: Hello, We're using asterisk with a z-plex 10 and a Zap tormentia card at a non-profit home for the aged. Sometimes when we dial out, our numbers are misintepreted with one number being detected twice. For instance 337-4411 becomes 337-7441 7's 4's and 0's seem to be particularly prone to this issue. Any ideas? I've tried turning down rxgain and txgain. A little debugging info: # uname -a Linux pbx 2.4.18 #1 Mon Aug 25 14:27:34 EDT 2003 i686 unknown Debian system 1.2Ghz Athalon Thunderbird Asterisk CVS-08/25/03-14:35:09, Copyright (C) 1999-2001 Linux Support Services, Inc. Do you hear a little static during the dialtone? Do you sometimes hear extra noise as you add on more active channels? If so it has to do with timing problems and slips. Asterisk will detect a DTMF then a static pop will signal the end of the DTMF, then asterisk hears the DTMF again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me
Ernest, Again, I really appreciate your help with this. Your solution looks like it requires two POTS lines -- am I misreading it? My goal is to have a call come in on a single POTS line and then have Asterisk try to track me down via the same POTS line (3 way calling.) Ben At 12:30 PM 9/17/2003 -0700, Ernest W. Lessenger wrote: At 06:48 PM 9/16/2003, you wrote: cell phone into the call (or my office number, etc.) I understand the selected numbers part of it, but not how to get it to use the three way. If I send it to Nufone first, I'm paying for a call to a local number (my cell) that I don't need to. This should work... [default] exten = s,1,Dial(Zap/3,20,t) ; This is your desk phone exten = s,2,Dial(Zap/2/1234567,20,t) ; This is your secondary POTS line calling your office exten = s,3,Dial(Zap/2/3217654,20,t) ; This is your secondary POTS line calling your cell phone ; I've never tried this one coming up, but I think it's worth a shot as it works just fine for local extensions exten = s,4,Dial(Zap/2/3217654Zap/3/3217654,20,t) ; This is your secondary and tertiary POTS lines calling your cell phone anbd office As long as none of these lines go to voicemail, they should fail over properly in order. You can also make it more complicated with time-based includes and gotos. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing out with the outgoing queue problem.
I got curious of this function and tried to summarize by reading your mails and looking into the source code. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out As you see, there are some commands available in the call file that I could not figure out. If you have figured these out, just add them to the Wiki page. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] number detection problem.
We had the same problem I found it useful to turn down the rxgain and txgain to -14 on those channels and make the T1 card the master clock source zaptel.conf span=1,1,0,esf,b8zs #for the T100P --- Note the 1 instead of 2 for the second parameter. span=1,1,0,esf,b8zs # For the T100PSteven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2003-09-26 at 15:09, Jon Hopper wrote: Hello, We're using asterisk with a z-plex 10 and a Zap tormentia card at a non-profit home for the aged. Sometimes when we dial out, our numbers are misintepreted with one number being detected twice. For instance 337-4411 becomes 337-7441 7's 4's and 0's seem to be particularly prone to this issue. Any ideas? I've tried turning down rxgain and txgain. A little debugging info: # uname -a Linux pbx 2.4.18 #1 Mon Aug 25 14:27:34 EDT 2003 i686 unknown Debian system 1.2Ghz Athalon Thunderbird Asterisk CVS-08/25/03-14:35:09, Copyright (C) 1999-2001 Linux Support Services, Inc.Do you hear a little static during the dialtone? Do you sometimes hearextra noise as you add on more active channels? If so i t has to do withtiming problems and slips. Asterisk will detect a DTMF then a static popwill signal the end of the DTMF, then asterisk hears the DTMF again.-- Steven Critchfield <[EMAIL PROTECTED]>___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] dialing out with the outgoing queue problem.
On Fri, 2003-09-26 at 16:36, Olle E. Johansson wrote: I got curious of this function and tried to summarize by reading your mails and looking into the source code. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out As you see, there are some commands available in the call file that I could not figure out. If you have figured these out, just add them to the Wiki page. Okay, I edited a few things there. BTW, I don't think the Manager API stuff needs to go together with the outgoing/call stuff. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy communication issue
Steve Lorimer wrote: I've setup a trial Asterisk install based on the RH8 install guide mentioned on this list. (Thanks, Andy!) I've configured two other working systems with SJPhone software for SIP. However, while I can call the phones, the communication is choppy. About every three or four seconds it cuts out briefly and then returns to normal. Perhaps someone can give me some pointers. My setups are described below: 1. Asterisk server 2.6 Athlon AMD / 512 MB Ram Using separate network card. Using built in sound card. (This should only cause problems for voice mail and prompts, correct?) 2. Client machine 1 1Ghz Celeron, 256mb Ram, etc. 3. Client machine 2 500 Mhz AMD, 96 mb Ram, etc. They are connected via a BayNetworks 350F-HD switch. (Fully capable, however, it does not have advanced packet routing/ prioritizing.) Is it practical to assume that the problem is in the switch? (ie. needing QoS for VOIP packets) Or perhaps I should be looking somewhere else to resolve this. I would appreciate any pointers or suggestions offered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Are the phones able to reinvite ( canreinvite=yes ) ? This for phone to phone conversation. Any Zaptel HW or ZTDummy loaded ? This for interact with * . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk license (fwd)
I can't understand why that was necessary. I'll made it simple. Asterisk is GPL ? I think yes Mysql seems that now is GPL. OpenH323 isn't GPL. So since Oh323 break GPL license , seems that h323 should be moved to the addons packages ,since in that way it also breaks * GPL license. Asterisk + mysql are ok together, since both are GPL. I don't see any problems with that. Seems that I cannot use Asterisk (GPL) + Oh323 (not GPL), cause the latter. Asterisk (GPL) + MySQL (GPL) should be ok. Any comments? Or I'm terribly blind? Matteo. Il sab, 2003-09-27 alle 00:50, Mark Spencer ha scritto: Just FYI, MySQL stuff has been pulled from Asterisk since apparently now the client libraries are under GPL and not LGPL (and thus are incompatible with OpenH323). You may check out the MySQL code under asterisk-addons, but you should not use both MySQL and OpenH323 (OpenSSL is also questionable) in the same Asterisk installation unless you downgrade your MySQL client libraries to a version that is before they GPL'd it. Sorry for the added inconvenience. I'm including the message from MySQL if any of you would like to try to encourage them to LGPL their client library again. Thanks! Mark -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
