RE: [Asterisk-Users] Small office requirements - Can this be done?
Steve Kennedy wrote: > On Mon, Mar 01, 2004 at 08:10:11PM -, Senad Jordanovic wrote: > >> Steve Kennedy wrote: >>> On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote: You could port your numbers to a licenced telco... Install SDSL (or even ADSL if you have a lot of faith in your current provider) and get all lines working throught SDSL.. :) You prabably should keep one BT line for fax and/or backup. >>> There's no such thing as a "licensed telco" anymore. Anyone >>> operating as a telco has to meet obligations set down in the >>> Communications Act, anyone meeting those obligations is a telco ... >>> as per EU dictate ... >> Well... >> Even better then... > > Of course SDSL has only limited coverage in the UK, and ADSL isn't > really sutable for more than one voice line (with any degree of "toll > grade" service). > > > Steve Have you tried running * box behind ADSL line, and push more than one call throught IAX and different codecs? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] having users in sql
Hi. If I want to have all my users (sip) in q mysql I've tried a few thingies.. but I didn't gett all the needed fields.. like nat, callerid, etc etc Is there a good way to solve this ? /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] having users in sql
On Tue, Mar 02, 2004 at 09:36:29AM +0100, Micke Andersson wrote: > > Hi. > > If I want to have all my users (sip) in q mysql > > I've tried a few thingies.. but I didn't gett all the needed fields.. > like nat, callerid, etc etc nat can be set globally in the [global] section (funnily enough.) It defaults to being off, so you need to explicitly enable it. Caller ID I thought was interesting as well. I was entertaining making the changes nessarcy so that could be used, but then the box I was using died.. > > Is there a good way to solve this ? > > /Mike > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
On Tue, 2004-03-02 at 06:35, Micke Andersson wrote: > Does anybody know or have good examples of using all functions in a 7960 > (SIP) http://voip-info.org/wiki-Asterisk+phone+cisco+79xx F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] having users in sql
On Tue, 2004-03-02 at 08:36, Micke Andersson wrote: > If I want to have all my users (sip) in q mysql > I've tried a few thingies.. but I didn't gett all the needed fields.. > like nat, callerid, etc etc http://voip-info.org/wiki-Asterisk+configuration+from+database F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues with chan_mgcp
I`m trying to implement a queue with MGCP channels. It does not detect that the channels is busy and does not try the next channel that is in the queue. When one of the members of the queue is busy, when he is talking to somebody I get this on the console: -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 1, sno: 0 I see the same messages when I put the channel in DND mode, when the channel is simply off-hook I see messages on the console that the channels is ringing. Even though I have a callwaiting=no in mgcp.conf. Does anybody have any success with chan_mgcp and queues? I have them working with SIP, but not MGCP Thanks -- Best regards, Evgeniy mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does it exist - DNS "TX" record?
When handed a URL type address for telephony, is there a DNS "TX" record (like MX but for telephone/Video) that could be looked up for an address to use to connect the call? I would like to have a "gateway server" (probably *) that anyone who knows the email address of a member of staff can use to connect to them with. If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Regards Chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: > I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from > Covad (in the US Southwest) and I have sustained 4 calls without a > problem. I prefer to use GSM over G.711to squeeze it down, but that is > my choice. I don't feel that call quality is substandard. That's the crunch (1.5/512) ... it's actually the 512 which is relevent. Virtually all DSL in the UK is a wholesale product from BT (they have about 2 million customers, Easynet who local loop unbundle may have 20,000, the rest of the providers maybe another 10,000 between them). All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 and 50:1, but actually a lot less than that), there are a few providers doing their own contention over BT's product. However the 256K upstream is still the limiting factor, so you can get one, and MAYBE two VoIP lines over it. If BT would up the upstream to 512, you could probaly get 4 out of it Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Steve Kennedy wrote: On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from Covad (in the US Southwest) and I have sustained 4 calls without a problem. I prefer to use GSM over G.711to squeeze it down, but that is my choice. I don't feel that call quality is substandard. That's the crunch (1.5/512) ... it's actually the 512 which is relevent. Virtually all DSL in the UK is a wholesale product from BT (they have about 2 million customers, Easynet who local loop unbundle may have 20,000, the rest of the providers maybe another 10,000 between them). All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 and 50:1, but actually a lot less than that), there are a few providers doing their own contention over BT's product. However the 256K upstream is still the limiting factor, so you can get one, and MAYBE two VoIP lines over it. If BT would up the upstream to 512, you could probaly get 4 out of it Steve On the UK DSL using G.711 you should easily get 2 concurrect calls, G.711 uses about 84k(incl overhead) in each direction, so 2 calls would be 168K (of the 256k) If you switch o GSM or iLBC you should get 6 concurrent calls, and if you were to use IAX2 trunking you could *maybe* squeeze another one.. Other codecs could offer even more but I haven't tested them.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
We do 4 per adsl with gsm every day. Matt - Original Message - From: "WipeOut" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 11:04 PM Subject: Re: [Asterisk-Users] Small office requirements - Can this be done? | Steve Kennedy wrote: | | >On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: | > | > | > | >>I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from | >>Covad (in the US Southwest) and I have sustained 4 calls without a | >>problem. I prefer to use GSM over G.711to squeeze it down, but that is | >>my choice. I don't feel that call quality is substandard. | >> | >> | > | >That's the crunch (1.5/512) ... it's actually the 512 which is relevent. | >Virtually all DSL in the UK is a wholesale product from BT (they have | >about 2 million customers, Easynet who local loop unbundle may have | >20,000, the rest of the providers maybe another 10,000 between them). | > | >All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 | >and 50:1, but actually a lot less than that), there are a few providers | >doing their own contention over BT's product. | > | >However the 256K upstream is still the limiting factor, so you can get | >one, and MAYBE two VoIP lines over it. If BT would up the upstream to | >512, you could probaly get 4 out of it | > | > | >Steve | > | > | > | On the UK DSL using G.711 you should easily get 2 concurrect calls, | G.711 uses about 84k(incl overhead) in each direction, so 2 calls would | be 168K (of the 256k) | | If you switch o GSM or iLBC you should get 6 concurrent calls, and if | you were to use IAX2 trunking you could *maybe* squeeze another one.. | | Other codecs could offer even more but I haven't tested them.. | | Later.. | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office requirements - Can this be done?
Matt Riddell wrote: > We do 4 per adsl with gsm every day. Who is your ADSL provider? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Matt Riddell wrote: We do 4 per adsl with gsm every day. Matt Like I said you should be able to easily do 6 when using GSM.. I have done 2 concurrent iLBC through IAX calls over a 64k ISDN link.. :) Later.. - Original Message - From: "WipeOut" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 11:04 PM Subject: Re: [Asterisk-Users] Small office requirements - Can this be done? | Steve Kennedy wrote: | | >On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: | > | > | > | >>I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from | >>Covad (in the US Southwest) and I have sustained 4 calls without a | >>problem. I prefer to use GSM over G.711to squeeze it down, but that is | >>my choice. I don't feel that call quality is substandard. | >> | >> | > | >That's the crunch (1.5/512) ... it's actually the 512 which is relevent. | >Virtually all DSL in the UK is a wholesale product from BT (they have | >about 2 million customers, Easynet who local loop unbundle may have | >20,000, the rest of the providers maybe another 10,000 between them). | > | >All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 | >and 50:1, but actually a lot less than that), there are a few providers | >doing their own contention over BT's product. | > | >However the 256K upstream is still the limiting factor, so you can get | >one, and MAYBE two VoIP lines over it. If BT would up the upstream to | >512, you could probaly get 4 out of it | > | > | >Steve | > | > | > | On the UK DSL using G.711 you should easily get 2 concurrect calls, | G.711 uses about 84k(incl overhead) in each direction, so 2 calls would | be 168K (of the 256k) | | If you switch o GSM or iLBC you should get 6 concurrent calls, and if | you were to use IAX2 trunking you could *maybe* squeeze another one.. | | Other codecs could offer even more but I haven't tested them.. | | Later.. | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office requirements - Can this be done ?
I've done a fare amount of analysis on codec bandwidth requirements and you should remember that you typically will require more bandwidth over ADSL than you would over any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711 channel rather than 86kb over straight PPP/HDLC based connections. Why I hear you ask ? The following calculations are based on G.711 PCM running at 20ms samples resulting in 200 byte packets (default for most codec implementations). 200 bytes G.711 packet + 8 bytes AAL5 overhead = 208 bytes 208 bytes fit in 5 cells of 48 bytes payload 5 cells are 265 bytes. VoIP over ATM AAL5MUX thus has an overhead of 21.51% VoIP G.711 conversation sends 50 packets per second. This uses 250 cells per second. This causes approximately 10 OAM5 cells to be sent over the duration. The total bitrate is thus (250 + 10) * 53 bytes * 8 bits = 110240 bits/second = 107.66Kbit/s Steve Kennedy wrote: >On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: > > > >>I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from >>Covad (in the US Southwest) and I have sustained 4 calls without a >>problem. I prefer to use GSM over G.711to squeeze it down, but that is >>my choice. I don't feel that call quality is substandard. >> >> > >That's the crunch (1.5/512) ... it's actually the 512 which is relevent. >Virtually all DSL in the UK is a wholesale product from BT (they have >about 2 million customers, Easynet who local loop unbundle may have >20,000, the rest of the providers maybe another 10,000 between them). > >All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 >and 50:1, but actually a lot less than that), there are a few providers >doing their own contention over BT's product. > >However the 256K upstream is still the limiting factor, so you can get >one, and MAYBE two VoIP lines over it. If BT would up the upstream to >512, you could probaly get 4 out of it > > >Steve > > > On the UK DSL using G.711 you should easily get 2 concurrect calls, G.711 uses about 84k(incl overhead) in each direction, so 2 calls would be 168K (of the 256k) If you switch o GSM or iLBC you should get 6 concurrent calls, and if you were to use IAX2 trunking you could *maybe* squeeze another one.. Other codecs could offer even more but I haven't tested them.. Later.. * 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Telecom (In New Zealand) :-) Matt - Original Message - From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 11:17 PM Subject: RE: [Asterisk-Users] Small office requirements - Can this be done? | Matt Riddell wrote: | > We do 4 per adsl with gsm every day. | | | Who is your ADSL provider? | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done ?
Low, Adam wrote: I've done a fare amount of analysis on codec bandwidth requirements and you should remember that you typically will require more bandwidth over ADSL than you would over any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711 channel rather than 86kb over straight PPP/HDLC based connections. Why I hear you ask ? The following calculations are based on G.711 PCM running at 20ms samples resulting in 200 byte packets (default for most codec implementations). 200 bytes G.711 packet + 8 bytes AAL5 overhead = 208 bytes 208 bytes fit in 5 cells of 48 bytes payload 5 cells are 265 bytes. VoIP over ATM AAL5MUX thus has an overhead of 21.51% VoIP G.711 conversation sends 50 packets per second. This uses 250 cells per second. This causes approximately 10 OAM5 cells to be sent over the duration. The total bitrate is thus (250 + 10) * 53 bytes * 8 bits = 110240 bits/second = 107.66Kbit/s So for us Dummies out here :) who just know it works. This would mean that if you had a 512/256 aDSL and a 256 ISDN connection you would be able to have more channels over the ISDN? David Steve Kennedy wrote: On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from Covad (in the US Southwest) and I have sustained 4 calls without a problem. I prefer to use GSM over G.711to squeeze it down, but that is my choice. I don't feel that call quality is substandard. That's the crunch (1.5/512) ... it's actually the 512 which is relevent. Virtually all DSL in the UK is a wholesale product from BT (they have about 2 million customers, Easynet who local loop unbundle may have 20,000, the rest of the providers maybe another 10,000 between them). All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 and 50:1, but actually a lot less than that), there are a few providers doing their own contention over BT's product. However the 256K upstream is still the limiting factor, so you can get one, and MAYBE two VoIP lines over it. If BT would up the upstream to 512, you could probaly get 4 out of it Steve On the UK DSL using G.711 you should easily get 2 concurrect calls, G.711 uses about 84k(incl overhead) in each direction, so 2 calls would be 168K (of the 256k) If you switch o GSM or iLBC you should get 6 concurrent calls, and if you were to use IAX2 trunking you could *maybe* squeeze another one.. Other codecs could offer even more but I haven't tested them.. Later.. * 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
Chris Lee wrote: If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Yes DNS has a TXT field, but in this case I think you're after ENUM.164, See the following URLs for details about ENUM: http://www.voip-info.org/wiki-ENUM http://www.voip-info.org/wiki-Asterisk+E164+Call+Routing however the problem with enum is the lack of wide spread deployment... Which has annoyed myself and others to try and think of a solution to deploy our own enum zone, but without conflicting with existing numbers, as using pots numbering would have 1 or 2 side effects, firstly people lie and abuse systems. Technically you could get people singing up with phone numbers for the white house. Using pots numbers could lead to serious privacy issues for individuals, businesses want people to find them, individuals might not want their phone number linked with their email, web address and so on and so forth and easily handed over to any and all that do a DNS lookup on their phone number. So what do you do? Well myself and a few friends bashed our heads together then started reading through a ton of ITU information about international direct dialling prefixes and found +882 is set aside for international bodies (companies) that provide communications such as satellite phones and the like. From that range ITU allocates sub prefixes 00 to 99 so far they're up to 34. Long story short this is the area most likely to be given to any sort of parallel system trying to get an enum service running. We've hacked up a few webpages to inject dns entries into mysql which power dns then spits out on request. To try this or even just to have a curious look go to: http://e164.freenetworks.org A lot of the work on this is for the benefit of community wireless groups, but is also useful for the wider internet community as well. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
Duane wrote: Chris Lee wrote: If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Yes DNS has a TXT field, but in this case I think you're after ENUM.164, See the following URLs for details about ENUM: http://www.voip-info.org/wiki-ENUM http://www.voip-info.org/wiki-Asterisk+E164+Call+Routing however the problem with enum is the lack of wide spread deployment... Which has annoyed myself and others to try and think of a solution to deploy our own enum zone, but without conflicting with existing numbers, as using pots numbering would have 1 or 2 side effects, firstly people lie and abuse systems. Technically you could get people singing up with phone numbers for the white house. Using pots numbers could lead to serious privacy issues for individuals, businesses want people to find them, individuals might not want their phone number linked with their email, web address and so on and so forth and easily handed over to any and all that do a DNS lookup on their phone number. So what do you do? Well myself and a few friends bashed our heads together then started reading through a ton of ITU information about international direct dialling prefixes and found +882 is set aside for international bodies (companies) that provide communications such as satellite phones and the like. From that range ITU allocates sub prefixes 00 to 99 so far they're up to 34. Long story short this is the area most likely to be given to any sort of parallel system trying to get an enum service running. We've hacked up a few webpages to inject dns entries into mysql which power dns then spits out on request. To try this or even just to have a curious look go to: http://e164.freenetworks.org A lot of the work on this is for the benefit of community wireless groups, but is also useful for the wider internet community as well. I am looking for a system that works like this (non phone number stuff): User enters SIP:// cslee-list at cybericom.co.uk into their sip "phone" The software does DNS lookup for TX record for cybericom.co.uk Uses TX record data to connect call to TX server belonging to Cybericom. TX server makes appropriate connections within Cybericom to cslee-list. Is something like this in the standards? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
Chris Lee wrote: Is something like this in the standards? Yes, enum can store any URI, SIP, IAX2, TEL, LDAP, HTTP, MAILTO... It's up to the client software as to how it handle the information returned... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office requirements - Can this be done ?
> So for us Dummies out here :) who just know it works. Yep, it sure does, I thought it was something people might find interesting. Its certainly been a challenging subject for me to try and provide reliable and high quality voice service over ADSL. In my experience it seems to depend a hell of a lot on the QoS deployed on the ATM network behind the DSLAM's. Obviously a single cell being dropped every 5 cells would effectively cause every G.711 IP packet to be lost. Here in Holland I ported my KPN (legacy incumbent) telephone number to my home VoIP service about 4 months ago. It has been running over a BBNED ADSL service and works great 99.9% of the time. Although during recently virus/worm outbreaks I have found people complain they hear my voice choppy, probably due to the contention of all the other ADSL connection upstreams as they propagate those viruses/worms. > This would mean that if you had a 512/256 aDSL and a 256 ISDN connection > you would be able to have more channels over the ISDN? Thats right, I am not aware of any ADSL providers that actually provide their stated service level at an IP layer rather than at the ATM layer but maybe they are out there ... The exact calculation depends on how your encapsulating IP over the 256k ISDN connection. I will assume your actually getting 4x B channels with either multi-link PPP (haven't calculated the overhead for this one) or a CSU/DSU converting to X.21/V.35 (preferable). You should be able to push 3 concurrent G.711 channels over that 256k ISDN service assuming 86Kbps per channel. > David Here's a little table I put together for our capacity planning team: G.711 over Ethernet = 95 Kbps per channel G.711 over IP/PPP = 86 Kbps per channel G.711 over ADSL/ATM = 108 Kbps per channel G.729 over Ethernet = 39 Kbps per channel G.729 over IP/PPP = 30 Kbps per channel G.729 over ADSL/ATM = 45 Kbps per channel * 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ext.conf for european variable length dialplan ?
Hello dialplan experts,
I am trying and browsing for days to find a solution for dialing out the
variable length phone numbers in Europe. We have area codes between 2
and 5 digits long and subscriber numbers between 4 and 8 digits making
pattern matching a pain.
For IP phones something like
exten => _0.,1,Dial(Zap/g1/${EXTEN:1},,t)
works well because the entire number comes in one setup message.
For "classical" phones people dial one digit after the other, the regex
matches after the first 0 and the outgoing ISDN channel starts dialing
dropping the rest of the digits.
My Cisco ATA 186 implements a timeout and/or the # as an 'enter' key
which would be kind of a second best solution but doesn't seem work with
ISDN. (BTW: Did somebody get this to work???)
The best solution would be overlap dialing until the PSTN switch says
'ringing' in the D-channel. Is this possible with zaptel channels?
May I ask what other european * users are using to handle this?
Thanks and regards,
Jan Baumann
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[Asterisk-Users] inband dtmf
Hi, I need some help with dtmf handling: I have sip phones configured to send dtmf via sip info, is it possible during an outside call (currently I have an i4l card) to convert those sip messages to real tones ? Tnx for any help ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
On Tue, Mar 02, 2004 at 10:04:23AM +, WipeOut wrote: > On the UK DSL using G.711 you should easily get 2 concurrect calls, > G.711 uses about 84k(incl overhead) in each direction, so 2 calls would > be 168K (of the 256k) It all to do with running IP/ATM (BT's ADSL is all over ATM) and how the multiplexing etc is done over the ATM network, such that with a 256K upstream you only really get 1/4 of the bandwidth for TOLL GRADE service. That doesn't mean you cant get more voice channels out of it, you can, but you'll start suffering on dropped packets etc. I'm sure some more knowledgable person can go into the maths behind IP/ATM and bitstuffing techniques to maintain latency and jitter. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
On Tue, Mar 02, 2004 at 10:40:25AM +, WipeOut wrote: > Like I said you should be able to easily do 6 when using GSM.. > I have done 2 concurrent iLBC through IAX calls over a 64k ISDN link.. :) ISDN is a 64K symmetrical system, ADSL is assymmetrical ... that's where the problems occur, and it's IP/ATM. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] flash button
Hi, Is there a way to control the flash timing in Asterisk? I'm using Siemens euroset 805S analog phones with Asterisk I can transfer a call just hitting a little slower on the "on-hook" button. The flash button is not working. I was trying to set in zapata.conf changing values of flash and rxflash and it did not work. Is there an other way to do this? best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call flip-flop with a phone connected to a TDM400P
Hello, I would like to do swicth alternatively two different calls from a phone connected to a TDM400P channel. I have found the way to do call transfert (http://www.voip-info.org/wiki-Asterisk+tips+zap+transfer) but it doesn't suits my needs. Instead of having a three call conferencing I would like to have a mean for bringing each call alternatively on and off. Any idea? 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
Thanks for the info Tim. -Matt TelCom Products International 2901 Frontage Road S Hwy 10E Moorhead, MN 56560 Phone# 218-422-9004 Fax# 218-422-9014 Support on MSN Messenger [EMAIL PROTECTED] - Original Message - From: "Tim Sailer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 01, 2004 9:29 PM Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage > On Mon, Mar 01, 2004 at 04:09:51PM -0600, Matt wrote: > > > > Hello, > > I found pebble linux, but asterisk is not packaged with it. > > Sorry, bad phrasing on my part. *Debian* has asterisk packaged for it, > so just 'apt-get install asterisk' within your pebble instance, and you > should be up and running with the basic setup. > > Tim > > -- > >< > >> Tim Sailer >< Coastal Internet, Inc. << > >> Network and Systems Operations >< PO Box 726 << > >> http://www.buoy.com >< Moriches, NY 11955 << > >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << > >< > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
In your DNS zone file for the domain you are using, put: _sip._udp SRV 0 0 5060sipproxy.yourdomain.com. sipproxy300 IN A 1.2.3.4 John Chris Lee wrote: When handed a URL type address for telephony, is there a DNS "TX" record (like MX but for telephone/Video) that could be looked up for an address to use to connect the call? I would like to have a "gateway server" (probably *) that anyone who knows the email address of a member of staff can use to connect to them with. If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Regards Chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
- Original Message - From: "Chris Lee" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 6:42 AM Subject: [Asterisk-Users] Does it exist - DNS "TX" record? > When handed a URL type address for telephony, is there a DNS "TX" record > (like MX but for telephone/Video) that could be looked up for an address > to use to connect the call? > I would like to have a "gateway server" (probably *) that anyone who > knows the email address of a member of staff can use to connect to them > with. > If the details of this server were in my DNS then anyone trying to call > someone at cybericom.co.uk could find the server to make the connection > with. Look here: http://www.voip-info.org/wiki-DNS+SRV ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done ?
David Uzzell wrote: So for us Dummies out here :) who just know it works. This would mean that if you had a 512/256 aDSL and a 256 ISDN connection you would be able to have more channels over the ISDN? David It all depends on what excapsulation your aDSL uses. It boils down to encapsulation overhead with "overhead" being the number of bits _in_addition_to_your_payload_ that are required to transit your payload. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office requirements - Can this be done?
Although there have also been times that with 2 concurrent calls we get dropped packets - Original Message - From: "Matt Riddell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, March 03, 2004 12:09 AM Subject: Re: [Asterisk-Users] Small office requirements - Can this be done? | Telecom (In New Zealand) | | :-) | | Matt | - Original Message - | From: "Senad Jordanovic" <[EMAIL PROTECTED]> | To: <[EMAIL PROTECTED]> | Sent: Tuesday, March 02, 2004 11:17 PM | Subject: RE: [Asterisk-Users] Small office requirements - Can this be done? | | | | Matt Riddell wrote: | | > We do 4 per adsl with gsm every day. | | | | | | Who is your ADSL provider? | | | | ___ | | Asterisk-Users mailing list | | [EMAIL PROTECTED] | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Ringback Tone when Dialing (outside caller to internal extension from auto-attendant)
I have been unsuccessful of yet to produce a ringback tone when trying as an outside caller to dial an internal extension from an auto-attendant. This is the scenario: 1. Outside caller dials main line that is going into an FXO card (X100P). 2. Auto-attendant answers 3. Outside caller dials an internal extension 4. Internal extensions are analog phones connected to SPA-2000 using SIP. In Step one, of course the caller gets a ringback tone until the auto-attendant calls (supplied by the PSTN). In Step 3, the extension rings but no ringback tone is supplied to the caller. Things I have tried thus far (_please read_). 1. Removing Answer application from auto-attendant 2. Adding “r” option to Dial application 3. Add Ringing application before dialing None of these options have worked. Please note that if I call from one internal extension to another, I get a ringback tone. As well if I call an outside line from an internal extension I get a ringback tone. I have searched the Asterisk Wiki, Asterisk Mailing List and google. Any help would be greatly appreciated.
Re: [Asterisk-Users] Any Gentoo Users Running ASTERISK had problems on recent emerge -u world?
G'day David, My * server runs Gentoo 1.4, and have recently started getting choppy audio -- usually during voice prompts from voicemail and my talking clock (digits dropped in the talking clock, and broken playback in voicemail: "press star for help, and pound to exit", for example, comes out "press r for help, and pound to-ound to ex-ound t-ound to exit". I thought it might have been a problem to do with a new NIC I installed a little while back, but from your report I have to wonder! The hardware is dual Athlon-MP 1600+, Tyan MB, 1GB RAM. Asterisk is CVS-02/03/04 (hmmm, month old, might have to update soon). On Mon, 1 Mar 2004, David Liu wrote: > So I stop gracefully and started asterisk again. It failed to start, > complaining about music on hold. With some further investigation, I > confirm it is to do with MPG123. I had to downgrade it to mpg123-0.59q > or 0.59r in order to start Asterisk successfully again. My mpg123 is still 0.59r-r3... Thanks for the tip, I will not be updating this... > Everything used to work until emerge -u worldperhaps next time I > shouldn't really do that until I test i on a test serveroh well... I have not done 'emerge world' recently, but have emerged a couple of things -- nothing that is associated with the running of *, or is even in use at present (there is an update to gcc and glibc available for example, but I have not taken them yet). On your machine, check out /var/log/emerge.log to see what packages got updated or installed after your 'emerge world'. There might be clues in there. I have noticed that the choppiness seems to occur when there is some contention for CPU resource, although I have not been able to work out what * is fighting against (load average doesn't change, nothing's getting logged). I'm chasing an odd occasional kernel oops on this machine also, may be related somehow... As this machine does all the local work for my office, I've been thinking about putting the * workload onto its own machine anyway -- I just don't trust Gentoo on the server anymore for some reason :-( What kernel is your machine running? I updated to 2.4.25_pre6-gss a little while ago, but I *thought* things were running okay after that. Prior to the current kernel I was on 2.4.23_pre8-gss-r2. Sorry for not having a clear answer, but hopefully some thoughts to help us work out what might be going on. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there an * based distro?
But the above, I mean is there a stipped down distro avalible, that comes with * ?
Re: [Asterisk-Users] RTP connection broken
Ahhh, you must have upgraded to firmware version 4.2. I had the same problem because I didn't find the new parameter that they added in this release for broken RTP connections. Here is how I fixed it: BROKENCONNECTIONEVENTTIMEOUT = 36 This makes the gateway drop the connection after an hour of no RTP packets. Hey, if you get RFC2833 DTMF bridging to work on that gateway, let me know. I currently have a bug report open on them because Asterisk doesn't seem to interoperate with the Audicodes in that respect. Ernest W. Lessenger wrote: We have an Audiocodes MP-108 that keeps dropping connections to voicemail after exactly ten seconds. All other calls are normal, and voicemail works fine from SIP devices other than the gateway. The reason given for dropping these calls is "RTP Connection Broken." I suspect that the gateway is sensing the lack of audio from Voicemail and is panicking. Any suggestions? --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming calls.
OK...first off thanks for the responses. However, I'm still having the
same sort of issue. I've looked through the two places that you
gentlemen suggested, and am still having the problem. Here is the error
message that I am receiving:
"-- Starting simple switch on 'Zap/1-1'
Mar 2 09:58:18 WARNING[1225991360]: pbx.c:1781 ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler
-- Hungup 'Zap/1-1'"
Now...here are the areas from my different config files.
>From Zaptel.conf
fxsks=1
loadzone = us
defaultzone=us
>From Zapata.conf
[channels]
language=us
context=generalstuff
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=no
;hidecallerid=no
callwaiting=yes
restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
channel => 1
>From extensions.conf
[generalstuff]
exten => _NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _NXX,2,Congestion
exten => _1NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _1NXXNXX,2,Congestion
exten => 4743399,1,Dial(SIP/mark,30)
include=uti
include=uti-mainst
include=ivr
Obviously I haven't included the entire files...just my basic stuff that
I'm dealing with right now. Just in case you forgotthis is
happening when I try to dial into my * box from the outside. As of
right now, I have it set up to do a basic forward to the soft-phone on
my desktop...that will all change, but first I just want to get inbound
calls working. Any suggestions would be greatly appreciated. Thanks.
Mark
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[Asterisk-Users] Disa & #
I want to be able to use DISA without sending the # sign to terminate the password. Actually, I rather be able to use a "fixed field of X digits" (i.e. 228 for the password, where if a match is found, the field is terminated and the call processes). Our system has been configured to act as a switch, basically, it accept's inbound calls, verifies account authorization, selects an outbound trunk, and routes the calls, think "call forwarding" and you'll have the basic concept. Having to allways use the # plays havoc with some of the Automatic route selection used in various PBX's (and even some automated dialers have problems processing the #) that I'm trying to route calls from. Essentially, I want these customers to be able to route calls through our asterisk without having to incur additional programming charges from their vendors. Being able to send an authorization string (or better yet being able to use their existing authorization string) that isn't terminated by # would make it a piece of cake to convert these accounts to our service. Has anyone had any experience passing authorization to DISA that isn't terminated with #? Has anyone got a recommendation as to a better way to achieve the same goal?
RE: [Asterisk-Users] Incoming calls.
Well, if you can't beat em...join em...
Since I couldn't figure out where the "s" extension and the "default"
context were coming from...I just put them there...
Maybe this was what you guys were trying to get me to do the whole time
lol...maybe not...anyway...it's alright now.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Tuesday, March 02, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Incoming calls.
OK...first off thanks for the responses. However, I'm still having the
same sort of issue. I've looked through the two places that you
gentlemen suggested, and am still having the problem. Here is the error
message that I am receiving:
"-- Starting simple switch on 'Zap/1-1'
Mar 2 09:58:18 WARNING[1225991360]: pbx.c:1781 ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler
-- Hungup 'Zap/1-1'"
Now...here are the areas from my different config files.
>From Zaptel.conf
fxsks=1
loadzone = us
defaultzone=us
>From Zapata.conf
[channels]
language=us
context=generalstuff
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=no
;hidecallerid=no
callwaiting=yes
restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
channel => 1
>From extensions.conf
[generalstuff]
exten => _NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _NXX,2,Congestion
exten => _1NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _1NXXNXX,2,Congestion
exten => 4743399,1,Dial(SIP/mark,30)
include=uti
include=uti-mainst
include=ivr
Obviously I haven't included the entire files...just my basic stuff that
I'm dealing with right now. Just in case you forgotthis is
happening when I try to dial into my * box from the outside. As of
right now, I have it set up to do a basic forward to the soft-phone on
my desktop...that will all change, but first I just want to get inbound
calls working. Any suggestions would be greatly appreciated. Thanks.
Mark
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[Asterisk-Users] Asterisk Passthrough
Hi all, I have a Wildcard TE410P and I want use them to add the asterisk capabilities to my office pbx. My problem is that I don't want to create panic in my collegue with a drastic change from "traditional" pbx to VoIP, so I want make a smooth change. My actual pbx is connected to the external world with 2 PRI interface, my idea is to insert asterisk in the middle, I want disconnect the two PRI, connect them to the asterisk and connect the asterisk with old pbx with a "cross cable". So, at the first step, my asterisk is simple a passthrough, but in the future I can change smoothly all my office phone and finally I can disconnect the old pbx. Ok, I'm at the first step, I have two problem: - First problem: what is the configuration of asterisk for "passthrough"? I have a good knowledge about SIP, IAX, and asterisk in general, I have build a working configuration with SIP phones (Grandstream Budget One) and asterisk with a PRI, but I don't know how I can configure the zaptel card for passthrough. - Second problem (is not a real problem): where I can find the diagram for a "cross cable" for PRI-to-PRI connection. I hope that my bad english and my bad brain are not a big problem for the understandig of my problem, if you don't understand something please ask. Thank you for help. Ciao Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: >My actual pbx is connected to the external world with 2 PRI interface, my >idea is to insert asterisk in the middle, I want disconnect the two PRI, >connect them to the asterisk and connect the asterisk with old pbx with a >"cross cable". >So, at the first step, my asterisk is simple a passthrough, but in the >future I can change smoothly all my office phone and finally I can >disconnect the old pbx. You've got two options here - you can use dacs in /etc/zaptel.conf to literally just cross-connect the PRIs. The two telco PRIs would come in on two of your ports, and would turn around and go back out on the other two. This happens in the zaptel module and doesn't make it up into asterisk. Your other option is to terminate the two PRIs into asterisk, and use asterisk to provide two PRIs into your PBX. This gives you access to the actual call routing. >Ok, I'm at the first step, I have two problem: >- First problem: what is the configuration of asterisk for "passthrough"? >I have a good knowledge about SIP, IAX, and asterisk in general, I have >build a working configuration with SIP phones (Grandstream Budget One) and >asterisk with a PRI, but I don't know how I can configure the >zaptel card for passthrough. >- Second problem (is not a real problem): where I can find the diagram for >a "cross cable" for PRI-to-PRI connection. If you're looking for a cable to go from the TE410P to your existing phone switch, you need a T1 crossover. Jared Smith has a good chart: http://www.jaredsmith.net/misc/cables/ Good luck, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody know about the Sayson 480i VoIP Screen Phone?
In looking for a screen phone to use with Asterisk I came across Sayson's new Aastra 480i SIP/MGCP/H323 screen phone. It looks just like the ADSI phone but offers dual 10/100 Ethernet jacks and apparently some kind of programmability much like the ADSI phone has, but using XML. Before I go pestering the nice people at Sayson, I thought I would see if anybody here knew anything about it - including details on the licensing/locking scheme. Also, any guess as to the price would be cool. I'm guessing it will be more than the equivalent ADSI. Link: http://www.sayson.com/product/voip_phone.htm Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] consultative call transfert with mgcp
Hello, I am faced to a problem with call transfert with a MGCP Phone. I use this to make a consultative call transfert: 1. send flash event 2. dial the number and speak with the other person 3. send flash event At this point asterisk tries to make a conference call with the three channels. My phone device doesn't support this. How should I do to make call transfert without trying to build a conference? Best 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP connection broken
At 07:07 AM 3/2/2004, you wrote: Ahhh, you must have upgraded to firmware version 4.2. I had the same problem because I didn't find the new parameter that they added in this release for broken RTP connections. Here is how I fixed it: BROKENCONNECTIONEVENTTIMEOUT = 36 That did it, thanks! Hey, if you get RFC2833 DTMF bridging to work on that gateway, let me know. I currently have a bug report open on them because Asterisk doesn't seem to interoperate with the Audicodes in that respect. I've tested what you describe with the MP-108, a SNOM 200 phone (2.03o firmware) and the most recent CVS of asterisk. No problems at all. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
> You've got two options here - you can use dacs in /etc/zaptel.conf to > literally just cross-connect the PRIs. The two telco PRIs would come in > on two of your ports, and would turn around and go back out on the other > two. This happens in the zaptel module and doesn't make it up into > asterisk. Can you elaborate on this? I have no mention of the term 'dacs' in /etc/zaptel.conf. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Steve Creel wrote: > Your other option is to terminate the two PRIs into asterisk, and use > asterisk to provide two PRIs into your PBX. This gives you access to the > actual call routing. Ok, my problem is exactly how I can do that? How can I say to asterisk "this port is connected to the world and the other is connected to my office telephon switch" (this is my main problem, I see something about groups but I'm not sure about the right configuration...)? How I can forward a call? It's simply an extension.conf rule? When I make the forward in this way (with extension.conf rule) asterisk make some work or is a simple passthrough from interfaces? I need that calls "from PRI to PRI" don't load the computer. I want to use all CPU to (future) SIP calls. Thank you for your reply. Ciao Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 - Signal - PBX connection
Hi all,
I've posted another message yesterday about the same problem. I will try
to be more detailed to get some help from the list:
I am connecting two PBX (PBX-A and PBX-B) using two * (*-A and *-B).
Asterisks are connected via IAX2. The PBXs area connected to each *
using an E1.
PBX-A <-- E1 --> *-A <-- IAX2 --> *-B <-- E1 --> PBX-B
When an extension from PBX-A calls an extension at PBX-B, PBX-A calls
*-A that calls *-B and then PBX-B. The problem is: if extension at B is
busy, *-A returns Normal Call Clear to PBX-A and PBX-A bills the call
normally. I contacted PBX support and they said me *-A should return a
different value to PBX-A (I think it is busy detected).
#
iax.conf at PBX-A:
[pbxb]
type=friend
auth=md5
secret=secret
bandwidth=low
disallow=all
allow=speex
#
extensions.conf at PBX-A:
[default]
exten => _.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
I tried to insert "|r" for ringback-only at Dial command and there is no
difference.
I search the mailing list and I found some emails about Call Signalling
and IAX protocol.
Any clue?
Daniel
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Re: [Asterisk-Users] consultative call transfert with mgcp
force all the users to a meetme extension ? On Tue, 2004-03-02 at 11:46, Daniel ANDRE wrote: > Hello, > > I am faced to a problem with call transfert with a MGCP Phone. I use > this to make a consultative call transfert: > 1. send flash event > 2. dial the number and speak with the other person > 3. send flash event > At this point asterisk tries to make a conference call with the three > channels. My phone device doesn't support this. How should I do to make > call transfert without trying to build a conference? > > Best regards, > > Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody know about the Sayson 480i VoIP Screen Phone?
> Link: http://www.sayson.com/product/voip_phone.htm When i asked in Dec it was to have a Q204 release some time in April us$250 was the target price, they have * setup to test it against in their North Vancouver, BC R&D Offices, the XML dialect was going to be their own , despite my best attempt to convince then of doing a xml rpc std. msg format, i have not talked then since this is the guy who was in charge of the xml dev James Ladan [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Anybody know about the Sayson 480i VoIP Screen Phone?
"Steven Sokol" <[EMAIL PROTECTED]> wrote: >In looking for a screen phone to use with Asterisk I came across Sayson's >new Aastra 480i SIP/MGCP/H323 screen phone. It looks just like the ADSI >phone but offers dual 10/100 Ethernet jacks and apparently some kind of >programmability much like the ADSI phone has, but using XML. > >Before I go pestering the nice people at Sayson, I thought I would see if >anybody here knew anything about it - including details on the >licensing/locking scheme. Also, any guess as to the price would be cool. >I'm guessing it will be more than the equivalent ADSI. I have been watching this phone for two or three months now. I don't think it is actually available yet. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] consultative call transfert with mgcp
William Suffill a écrit: force all the users to a meetme extension ? I don't really understand. On Tue, 2004-03-02 at 11:46, Daniel ANDRE wrote: Hello, I am faced to a problem with call transfert with a MGCP Phone. I use this to make a consultative call transfert: 1. send flash event 2. dial the number and speak with the other person 3. send flash event At this point asterisk tries to make a conference call with the three channels. My phone device doesn't support this. How should I do to make call transfert without trying to build a conference? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] Re: Anybody know about the Sayson 480i VoIP Screen Phone?
480i is said to be released sometime in April Retail will be aprox. $250 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Tuesday, March 02, 2004 12:13 PM To: Asterisk Users Subject: [Asterisk-Users] Re: Anybody know about the Sayson 480i VoIP Screen Phone? "Steven Sokol" <[EMAIL PROTECTED]> wrote: >In looking for a screen phone to use with Asterisk I came across >Sayson's new Aastra 480i SIP/MGCP/H323 screen phone. It looks just >like the ADSI phone but offers dual 10/100 Ethernet jacks and >apparently some kind of programmability much like the ADSI phone has, but using XML. > >Before I go pestering the nice people at Sayson, I thought I would see >if anybody here knew anything about it - including details on the >licensing/locking scheme. Also, any guess as to the price would be cool. >I'm guessing it will be more than the equivalent ADSI. I have been watching this phone for two or three months now. I don't think it is actually available yet. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anybody know about the Sayson 480i VoIP Screen Phone?
Lol, I didn't see this email. Oh well I posted this info as well. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Tuesday, March 02, 2004 12:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Anybody know about the Sayson 480i VoIP Screen Phone? > Link: http://www.sayson.com/product/voip_phone.htm When i asked in Dec it was to have a Q204 release some time in April us$250 was the target price, they have * setup to test it against in their North Vancouver, BC R&D Offices, the XML dialect was going to be their own , despite my best attempt to convince then of doing a xml rpc std. msg format, i have not talked then since this is the guy who was in charge of the xml dev James Ladan [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VTGO-PG and IPP200
First, thank you to whomever it was that pointed me towards the IPP200. It works great, both under Windows with X-lite and Linux with iaxcomm. It took me about 3 minutes on each to get it configured and working. Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd dearly love a softphone with MWI... Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Voicemenu question
Hi
I can get the Voicemenu stuff working OK but am unable to switch to a
context where the incoming PSTN caller is able to enter SIP number
(after promting) and have this forwarded to a proxy.What I am trying to
do is give a PSTN caller the choice between voicemail, local extension
or remote SIP user.
Below is an extract from my extensions.conf, clearly this does not work.
Any hint would be most appreciated.
Thanks
Brian
[incoming]
exten => s,1,Answer
exten => s,2,Background(brian-ivr)
exten => 6,1,Voicemail,u5152
exten => 7,1,Goto,pstn-to-sip|s|1
exten => 8,1,Dial,Zap/2
;
[pstn-to-sip]
exten =>s,1,Background(pstn-to-sip); ask user to enter sip number
exten =>9,_9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten =>8,_8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten =>7,_7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
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RE: [Asterisk-Users] VTGO-PG and IPP200
Hmmm. I thought it was just that I didn't have X-Lite and Asterisk configured correctly and I've been searching thru docs trying to figure out how to get a MWI working! Does X-Pro have a MWI? Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Tuesday, March 02, 2004 9:54 AM To: Asterisk Users Subject: [Asterisk-Users] VTGO-PG and IPP200 First, thank you to whomever it was that pointed me towards the IPP200. It works great, both under Windows with X-lite and Linux with iaxcomm. It took me about 3 minutes on each to get it configured and working. Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd dearly love a softphone with MWI... Tim -- ><<< ><< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << ><<< ><< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Voicemenu question
Brian,
You need to put an include => default in your incoming context.
some samples here http://www.codepipe.com/id25.htm
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Mulligan
Sent: 02 March 2004 17:58
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Voicemenu question
Hi
I can get the Voicemenu stuff working OK but am unable to switch to a
context where the incoming PSTN caller is able to enter SIP number
(after promting) and have this forwarded to a proxy.What I am trying to
do is give a PSTN caller the choice between voicemail, local extension
or remote SIP user.
Below is an extract from my extensions.conf, clearly this does not work.
Any hint would be most appreciated.
Thanks
Brian
[incoming]
exten => s,1,Answer
exten => s,2,Background(brian-ivr)
exten => 6,1,Voicemail,u5152
exten => 7,1,Goto,pstn-to-sip|s|1
exten => 8,1,Dial,Zap/2
;
[pstn-to-sip]
exten =>s,1,Background(pstn-to-sip); ask user to enter sip number
exten =>9,_9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten =>8,_8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten =>7,_7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
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Re: [Asterisk-Users] Record Application
Scott, reply is at the bottom. > I ran into a small issue with the Record app. If I specify a silence > detection period the application makes the recording and hangs > up,instead of continuing with the dial-plan as it should. Now if I > dont specify a silence detection period an use only the # to end all > recording sessions then everything works great, I make the recording, > press # and the recording finishes and the system continues with the > dial plan. > > With silence: > > exten => 205,2,Record(/tmp/asterisk-recording:gsm|5) > > Without Silence: > > exten => 205,2,Record(/tmp/asterisk-recording:gsm) > > I also receive this error message when I have the silence specified. > > NOTICE[475151]: Unable to find a path from ULAW to UNKN > WARNING[475151]: Unable to restore read format on > '[EMAIL PROTECTED]:4569]/2' I'm seeing similar behavior using the RECORD FILE command in AGI. If a silence timeout is specified, I get the same messages (replace 'ULAW' with 'GSM') when the recording times out, whether the timeout is due to maximum record time exceeded or the maximum silence exceeded. If the user ends the recording with # the errors do not occur, even when there is a silence parameter. I have only tried this over IAX (with no dummy timer), and I was just settling down to install a SIP client to try when I read your message. The good news for you is that when this happens in AGI it does not appear to have any obvious adverse effects. Asterisk spits back a successful result string and continues with the script. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VTGO-PG and IPP200
IAX Phone has MWI and, with a small change in your Asterisk, can give you a count of messages waiting. Plus it has hook-switch integration with the IPP200. Thanks, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Tim Sailer > Sent: Tuesday, March 02, 2004 11:54 AM > To: Asterisk Users > Subject: [Asterisk-Users] VTGO-PG and IPP200 > > First, thank you to whomever it was that pointed me towards the > IPP200. It works great, both under Windows with X-lite and Linux > with iaxcomm. It took me about 3 minutes on each to get it > configured and working. > > Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd > dearly love a softphone with MWI... > > Tim > > -- > >< > >> Tim Sailer >< Coastal Internet, Inc. << > >> Network and Systems Operations >< PO Box 726 << > >> http://www.buoy.com >< Moriches, NY 11955 << > >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << > >< > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New to T-1/Channel Bank hardware -- help?
Rob Fugina <[EMAIL PROTECTED]> wrote: >I'm considering a small office setup with at least 12 extensions. >Seems (as has been stated in previous threads) that for the FXS ports, a >T100P and a channel bank could be the most cost-effective way to do this. I'm not so sure about that. 6 Sipura SPA-2000s will cost you about US $540. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
Will it do dtmf out of band with the IPP200 ? On Tuesday 02 of March 2004 19:11, Steven Sokol wrote: > IAX Phone has MWI and, with a small change in your Asterisk, can give you a > count of messages waiting. Plus it has hook-switch integration with the > IPP200. > > Thanks, > > Steve > > Steven Sokol > Owner/Manager > Sokol & Associates, LLC > > Phone: 816.822.1807 > IaxTel: 700.613.9004 > Web:http://www.sokol-associates.com > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Tim Sailer > > Sent: Tuesday, March 02, 2004 11:54 AM > > To: Asterisk Users > > Subject: [Asterisk-Users] VTGO-PG and IPP200 > > > > First, thank you to whomever it was that pointed me towards the > > IPP200. It works great, both under Windows with X-lite and Linux > > with iaxcomm. It took me about 3 minutes on each to get it > > configured and working. > > > > Second, has anyone tried the VTGO-PC (any version) from ipblue.com? I'd > > dearly love a softphone with MWI... > > > > Tim > > > > -- > > > > > > > >< > > >> > > >> Tim Sailer >< Coastal Internet, Inc. > > >> << Network and Systems Operations >< PO Box 726 > > >> << http://www.buoy.com >< Moriches, NY 11955 > > >> << [EMAIL PROTECTED] >< (631) 399-2910 (888) > > >> 924-3728 << > > >> > > > > > >< > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls not hanging up.
I really do not think this is an asterisk problem, but I was not sure were else to look for support. I am using an asterisk setup to phone out church services to people via the meetme app. The way I have it setup, you phone in, give the system your telephone number and it calls you back via nufone or voicepulse over IAX2. The complaint I'm getting from a few people is that when they hang up their phones, they still cannot get dialtone for a while. Two people said last night that even 20 seconds after they hung up their phones, when they picked up again, they still did not have a dial tone. I'm not sure when it came back. For most people it works fine. Any suggestions? I don't think it is their phones because it worked fine for both of them other times. Then again, I don't know what it could be besides their phones. Darren Wiebe [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button
Osvaldo Mundim wrote: > Is there a way to control the flash timing in Asterisk? I'm using > Siemens euroset 805S analog phones with Asterisk I can transfer a call > just hitting a little slower on the "on-hook" button. The flash button > is not working. I had exactly the same problem with exactly the same phone ;-) When I pressed the flash button, Asterisk interpreted the event as a pulse-dialed "1". I looked into zaptel.h and found the following line: #define ZT_MAXPULSETIME (150 * 8)/* 150 ms maximum */ Every pulse shorter than that value is treated as a pulse-dial pulse. Since all my phones seem to generate 100 ms flash pulses, I reduced the "150" to "75" and recompiled Zaptel & Asterisk. The problem is solved - and pulse-dialing still works. regards, Pascal Le Bail, Vienna, Austria, Europe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
On Tue, Mar 02, 2004 at 09:59:55AM -0800, Ed Rubright wrote: > Hmmm. I thought it was just that I didn't have X-Lite and Asterisk > configured correctly and I've been searching thru docs trying to figure out > how to get a MWI working! Does X-Pro have a MWI? Not that I can find from the manual or web page. I have MWI working with the BT-100 hardphone, but using the same SIP setup for the extension, I get nothing from X-Lite. I'm looking for a softphone (SIP I guess. I don't see MWI indicated for IAX) that will support MWI and Hold, nothing fancy. $50-100 is about the range I'll look for, because the hardphone are more cost effective at that point. Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
On Tue, Mar 02, 2004 at 12:11:22PM -0600, Steven Sokol wrote: > IAX Phone has MWI and, with a small change in your Asterisk, can give you a > count of messages waiting. Plus it has hook-switch integration with the > IPP200. I'd like to try it, but I'm unable to get it to even start on my Lose2k system... Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VTGO-PG and IPP200
The VTGO-PC appears to be Cisco Call Manager only or did you find somewhere on there site that they have a SIP version? If it is Call Manager only...how far along is Asterisk support for Call Manager? I looked on the Wiki pages and saw mentioned the 2 implementations of the "skinny" protocol, but nothing about its current feature state. Thanks in advance, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Tuesday, March 02, 2004 10:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VTGO-PG and IPP200 On Tue, Mar 02, 2004 at 09:59:55AM -0800, Ed Rubright wrote: > Hmmm. I thought it was just that I didn't have X-Lite and Asterisk > configured correctly and I've been searching thru docs trying to > figure out how to get a MWI working! Does X-Pro have a MWI? Not that I can find from the manual or web page. I have MWI working with the BT-100 hardphone, but using the same SIP setup for the extension, I get nothing from X-Lite. I'm looking for a softphone (SIP I guess. I don't see MWI indicated for IAX) that will support MWI and Hold, nothing fancy. $50-100 is about the range I'll look for, because the hardphone are more cost effective at that point. Tim -- ><<< ><< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << ><<< ><< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Garbled Faxes
"Jim Sneeringer" <[EMAIL PROTECTED]> wrote: >My incoming and outgoing faxes are garbled and sometimes get disconnected. >I remember reading somewhere that I should use u-law for faxes, but I don't >know how to do that. The fax is connected to a Digium FXS card and the >calls come in or go out on a Digium FXO card. > >Can anyone tell me how to fix this? If the FXS and FXO cards are in the same box, then I don't think there is any issue with codec selection. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button
That worked perfectly!!! I've adjusted the ZT_MINPULSETIME to 4ms and its working fine! thank you best regards Osvaldo On Mar 2, 2004, at 3:21 PM, Pascal Le Bail wrote: Osvaldo Mundim wrote: Is there a way to control the flash timing in Asterisk? I'm using Siemens euroset 805S analog phones with Asterisk I can transfer a call just hitting a little slower on the "on-hook" button. The flash button is not working. I had exactly the same problem with exactly the same phone ;-) When I pressed the flash button, Asterisk interpreted the event as a pulse-dialed "1". I looked into zaptel.h and found the following line: #define ZT_MAXPULSETIME (150 * 8)/* 150 ms maximum */ Every pulse shorter than that value is treated as a pulse-dial pulse. Since all my phones seem to generate 100 ms flash pulses, I reduced the "150" to "75" and recompiled Zaptel & Asterisk. The problem is solved - and pulse-dialing still works. regards, Pascal Le Bail, Vienna, Austria, Europe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] add g.729 license
Hi, I already have one g.729 license on *. Could anyone tell me if I want to add a few more, can I just buy these online and follow their installation instruction, and * will add these addtional licenses? Or this will invalidate my current license? thanks ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E911 support
I haven't looked into it, but either * or the AS5350 gateway that I use sees the "Anonymous" text and sets the appropriate flags. > -Original Message- > From: John Fraizer [mailto:[EMAIL PROTECTED] > Sent: Thursday, February 26, 2004 3:32 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] E911 support > > Steve Dolloff wrote: > > I have the following in my sip.conf entries: > > > > callerid="Anonymous" <8885551212> > > > > This still passes the number for 911, but flags the call as private. I > > believe this will meet your requirements. > > > > Stephen > > OK. I was under the impression that the PSAP got their information based > on > ALI/ANI and not from CLID. Are you telling me that they're looking at > CLID? > > Also, at least in the testing I've done, the text portion of the CLID > string > is ignored by the telco. They only look at the number and generate the > text > based on what is in their database. IE; If I tell my asterisk server to > set > my callerID to "test" and call someplace, What I get on > the > CLID display of the phone I dial is "John Fraizer" and my home number. > > Since Powell has stated that we must provide E911 services, I am wondering > what precisely is going to have to be done to do so with Asterisk. > Routing > the call to the PSAP when someone dials 911 is the easy part. Sending all > of the information they want/need (much more than just CLID and something > that is regulated) is an alltogether different story. > > John > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM[RBL] [Asterisk-Users] add g.729 license
I was snooping through the archives and I think you have to email digium and give them your old license number. I think then you can buy more and they can send you an updated license which covers all of them. I would check with digium, but I really think you will lose your present license if you overwrite it. Darren Wiebe [EMAIL PROTECTED] Ron McMillin wrote: Hi, I already have one g.729 license on *. Could anyone tell me if I want to add a few more, can I just buy these online and follow their installation instruction, and * will add these addtional licenses? Or this will invalidate my current license? thanks ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] add g.729 license
You can purchase more licenses from diguim. Asterisk will overwrite your previous license. You would have to get someone from diguim to combine your 2 different licenses into one. Like if you had 2 license and want to upgrade to 4. You would have to purchase another 2 channel license and have both the license combined. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ron McMillin Sent: Tuesday, March 02, 2004 1:58 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] add g.729 license Hi, I already have one g.729 license on *. Could anyone tell me if I want to add a few more, can I just buy these online and follow their installation instruction, and * will add these addtional licenses? Or this will invalidate my current license? thanks ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Garbled Faxes
Thanks. They are in the same box. Anyone have any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith Sent: Tuesday, March 02, 2004 12:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Garbled Faxes "Jim Sneeringer" <[EMAIL PROTECTED]> wrote: >My incoming and outgoing faxes are garbled and sometimes get disconnected. >I remember reading somewhere that I should use u-law for faxes, but I don't >know how to do that. The fax is connected to a Digium FXS card and the >calls come in or go out on a Digium FXO card. > >Can anyone tell me how to fix this? If the FXS and FXO cards are in the same box, then I don't think there is any issue with codec selection. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 not ringing.
I was just wondering if anyone has had this situation...or one similar to it. I've got a Sipura SPA 2000. After hooking it up and configuring it with my * box, it has worked well. From both lines we are able to dial out at any point in time. However after a few minutes (5-10 usually) the Sipura will stop sending a ringer signal to the phones. We can still dial, * shows that it is ringing those SIP clients, and both lines are still shown as being "registered". Everything will work fine for the first 5-10 minutes after being "rebooted"...however after that there is silence. Even if I dial the line from my cell phone and pick up the telephone that I am dialing...there is dial-tone...therefore it seems that the Sipura is just not passing those incoming calls after a short time period. If you have any idea why this is happening I'd sure appreciate hearing your thoughts. Thanks Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
On Tue, Mar 02, 2004 at 10:40:24AM -0800, Ed Rubright wrote: > The VTGO-PC appears to be Cisco Call Manager only or did you find somewhere > on there site that they have a SIP version? I'm bugging them about the SIP, but so far, no answer. They claim it's a software close of the Cisco hardphone, so... > If it is Call Manager only...how far along is Asterisk support for Call > Manager? I looked on the Wiki pages and saw mentioned the 2 implementations > of the "skinny" protocol, but nothing about its current feature state. I'm not sure what state it's in. Or how to configure it for that matter. I may grab the eval copy and try to crash * with it... :) Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Andrew Kohlsmith wrote: >> You've got two options here - you can use dacs in /etc/zaptel.conf to >> literally just cross-connect the PRIs. The two telco PRIs would come in >> on two of your ports, and would turn around and go back out on the other >> two. This happens in the zaptel module and doesn't make it up into >> asterisk. > >Can you elaborate on this? I have no mention of the term 'dacs' >in /etc/zaptel.conf. According to asterisk-cvs, on October 30, 2003, dacs support was added: "Add DACS functionality to zaptel for cross connecting channels" zaptel.conf.sample is appropriately documented: "dacs" The zaptel driver cross connects the channels starting at the channel number listed at the end, after a colon If I wanted to cross connect the first span to the second: dacs = 1-24:25 If I want to cross connect just channel 3 to channel 27: dacs = 3:27 I imagine (though haven't tried it), you can use: dacs = 1,3-5:25 to take channels 1,3,4,5 and put them on 25,26,27,28 One note: you can only use dacs on T1/E1 spans, not the pci fxs/fxo cards. Hope that helps... Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Passthrough
On Tue, 2 Mar 2004, Emanuele Laface wrote: >On Tue, 2 Mar 2004, Steve Creel wrote: > >> Your other option is to terminate the two PRIs into asterisk, and use >> asterisk to provide two PRIs into your PBX. This gives you access to the >> actual call routing. > >Ok, my problem is exactly how I can do that? >How can I say to asterisk "this port is connected to the world and the >other is connected to my office telephon switch" (this is my main >problem, I see something about groups but I'm not sure about the right >configuration...)? Don't try to map port to port - you're making your problem more complex than it needs to be. Let asterisk do some call routing for you. You've got an incoming call with dialed number identification. Write an asterisk extension rule to handle it... Should asterisk send it out on a specific channel? Should it be sent out to one channel out of a group? For example, say your incoming number is 12345. You've connected the telco PRIs to spans 1 and 2, and your PRIs to the existing PBX are spans 3 and 4. The telco channels are all in group 1, the channels to the PBX are in group 2. [incoming] exten => 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX Now you have asterisk switching the call instead of cross connecting the ports. >How I can forward a call? It's simply an extension.conf rule? Yes. >When I make the forward in this way (with extension.conf rule) asterisk >make some work or is a simple passthrough from interfaces? Yes, it's some switching/callsetup work, but no codec translation, which is by far your biggest CPU consumer. >I need that calls "from PRI to PRI" don't load the computer. >I want to use all CPU to (future) SIP calls. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
At 10:25 PM +1100 3/2/04, Duane wrote: Chris Lee wrote: If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Yes DNS has a TXT field, but in this case I think you're after ENUM.164, See the following URLs for details about ENUM: http://www.voip-info.org/wiki-ENUM http://www.voip-info.org/wiki-Asterisk+E164+Call+Routing however the problem with enum is the lack of wide spread deployment... Which has annoyed myself and others to try and think of a solution to deploy our own enum zone, but without conflicting with existing numbers, as using pots numbering would have 1 or 2 side effects, firstly people lie and abuse systems. Technically you could get people singing up with phone numbers for the white house. Using pots numbers could lead to serious privacy issues for individuals, businesses want people to find them, individuals might not want their phone number linked with their email, web address and so on and so forth and easily handed over to any and all that do a DNS lookup on their phone number. So what do you do? Well myself and a few friends bashed our heads together then started reading through a ton of ITU information about international direct dialling prefixes and found +882 is set aside for international bodies (companies) that provide communications such as satellite phones and the like. From that range ITU allocates sub prefixes 00 to 99 so far they're up to 34. Long story short this is the area most likely to be given to any sort of parallel system trying to get an enum service running. We've hacked up a few webpages to inject dns entries into mysql which power dns then spits out on request. To try this or even just to have a curious look go to: http://e164.freenetworks.org A lot of the work on this is for the benefit of community wireless groups, but is also useful for the wider internet community as well. -- Best regards, Duane Are you aware of the +878 "country" code and the UPT (Universal Personal Telecommunications) project? http://www.visionng.org/index.htm http://www.visionng.org/enum/Request_for_Temporary_Assignment_of_UPT_Numbers.pdf&e=7413 As an example, I can be reached on +878102843336600 which is available via an e164.arpa. ENUM lookup. This is only now coming out of "beta" mode and into real production, but you may find it more reasonable to use a number range that has been correctly allocated rather than simply hijacking one that may in the future not be routed in ways that you expect or desire. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VTGO-PG and IPP200
On Tue, 2 Mar 2004 13:22:05 -0500, Tim Sailer <[EMAIL PROTECTED]> wrote: >On Tue, Mar 02, 2004 at 09:59:55AM -0800, Ed Rubright wrote: >> Hmmm. I thought it was just that I didn't have X-Lite and Asterisk >> configured correctly and I've been searching thru docs trying to figure out >> how to get a MWI working! Does X-Pro have a MWI? > >Not that I can find from the manual or web page. I have MWI working >with the BT-100 hardphone, but using the same SIP setup for the extension, >I get nothing from X-Lite. I'm looking for a softphone (SIP I guess. I don't >see MWI indicated for IAX) that will support MWI and Hold, nothing fancy. >$50-100 is about the range I'll look for, because the hardphone are more >cost effective at that point. IAX Phone (Hi, Steve!) has a MWI indicator http://www.sokol-associates.com/IaxPhone.htm > >Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN 4.7 and 5.0 Questions ..
Hi, Did someone tell-me why in MSN 4.7, when i connect my users does't appear online in the list , but when i put my voicemail contact in asterisk appears ? There is a phonepad in MSN 5.0 ? Like in 4.7 MSN 5.0 use the password very well with Asterisk ! But has the same problem that people never appears online!! (Even voicemail dont) Anyone have a solution for that ? Put people when connect over 4.7 or 5.0 to appears online !? Thanks alot! Oh yes, there is anyway to put MSN 4.7 to use password ? I try this : [carlos] type=friend insecure=yes username=carlos secret=teste host=dynamic qualify=1000 mailbox=300 Using with 5.0 works very well but i must disable the secret to use with 4.7. Both cases people appear offline when connect with asterisk... :( Thanks alot! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRTP: followup
I have found few VoIP clients that support encryption. The only one that comes to mind is the Zultys devices (they have a softphone and a hardphone that support SRTP.) I spoke with them as recently as today, asking if they'd be interested in supplying patches for their AES-based SRTP encryption implementation to Asterisk, and they said that they didn't have the ability to put those patches together at the moment, but they'd be happy testing functionality with their devices and Asterisk once implemented. Has anyone done any work with SRTP and Asterisk that could become public? I've still got a number of clients who aren't moving towards VoIP due to security concerns with end-user stations. There are now quite a few AES-specific files stuffed into the Asterisk distribution, and a quick "grep" doesn't show anything using them. Hopefully this means that we're almost at the edge of making SRTP a reality... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not ringing.
Did you try to upgrade the firmware?, some issues we saw with rtp stream, went away after a firmware upgrade. http://www.sipura.com -SamW -Original Message- From: Mark Messmore, Technical Support, University Telcom Inc. [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 1:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura 2000 not ringing. I was just wondering if anyone has had this situation...or one similar to it. I've got a Sipura SPA 2000. After hooking it up and configuring it with my * box, it has worked well. From both lines we are able to dial out at any point in time. However after a few minutes (5-10 usually) the Sipura will stop sending a ringer signal to the phones. We can still dial, * shows that it is ringing those SIP clients, and both lines are still shown as being "registered". Everything will work fine for the first 5-10 minutes after being "rebooted"...however after that there is silence. Even if I dial the line from my cell phone and pick up the telephone that I am dialing...there is dial-tone...therefore it seems that the Sipura is just not passing those incoming calls after a short time period. If you have any idea why this is happening I'd sure appreciate hearing your thoughts. Thanks Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record Application
> > NOTICE[475151]: Unable to find a path from ULAW to UNKN > > WARNING[475151]: Unable to restore read format on > > '[EMAIL PROTECTED]:4569]/2' > > I'm seeing similar behavior using the RECORD FILE command in AGI. If > a silence timeout is specified, I get the same messages (replace > 'ULAW' with 'GSM') when the recording times out, whether the timeout > is due to maximum record time exceeded or the maximum silence > exceeded. If the user ends the recording with # the errors do not > occur, even when there is a silence parameter. > > I have only tried this over IAX (with no dummy timer), and I was just > settling down to install a SIP client to try when I read your message. I have now tested it with over SIP using X-Lite, and I can't get the error to occur. I never bothered reporting this because it didn't break anything for me and I haven't gotten around to compiling with the dummy timer, but you may want to add it to the bug tracker. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not ringing.
Yeah...I upgraded to 1.0.31 (whatever the current is...I installed it less than 1 week ago)... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SamW Sent: Tuesday, March 02, 2004 3:19 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura 2000 not ringing. Did you try to upgrade the firmware?, some issues we saw with rtp stream, went away after a firmware upgrade. http://www.sipura.com -SamW -Original Message- From: Mark Messmore, Technical Support, University Telcom Inc. [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 1:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura 2000 not ringing. I was just wondering if anyone has had this situation...or one similar to it. I've got a Sipura SPA 2000. After hooking it up and configuring it with my * box, it has worked well. From both lines we are able to dial out at any point in time. However after a few minutes (5-10 usually) the Sipura will stop sending a ringer signal to the phones. We can still dial, * shows that it is ringing those SIP clients, and both lines are still shown as being "registered". Everything will work fine for the first 5-10 minutes after being "rebooted"...however after that there is silence. Even if I dial the line from my cell phone and pick up the telephone that I am dialing...there is dial-tone...therefore it seems that the Sipura is just not passing those incoming calls after a short time period. If you have any idea why this is happening I'd sure appreciate hearing your thoughts. Thanks Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: codec negotiation prob solved
I agree, that * codec negotiation is buggy, there must be some mechanism to give priority to pass through without trying to codec translate. Codec translation need lot of CPU and can deteriorate quality by its nature. Some developer sheding some light on this is buggy codec translation is very appreciated. - SamW -Original Message- From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Friday, February 20, 2004 12:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved I have the same problem, most carriers out there deal with both g723.1 or g729. During passing through via Asterisk, carrier customers will send us calls broadcasting both codecs with one having priority over the other, the way it is supposed to work is that asterisk will try to negotiate the top priority codec first with the terminating endpoint, assuming that the originating endpoint broadcasts g729 as first priority and then g723.1, Asterisk should take g729 and try to negotiate with terminating endpoint and if the terminating takes g729, then the call should be patched and bridged, but if the terminating endpoint takes ONLY g723.1, then Asterisk should then go back and take g723.1 (which is the second priority as per the originating endpoint) and bridge the call through. However, the way Asterisk is doing it is if I allow both g723.1 and g729, then if the originating endpoint broadcasts both codecs and the terminating endpoint only allows g723.1, then the call will not go through and it will say no path from g729 to ., and calls will not go through. Summing up, if originating gateway allows both g723.1 and g729 , Asterisk being the pass-through entity, allows both codecs, and the terminating gateway allows ONLY g723.1, the calls will not go through which is certainly a bug in the asterisk. I wonder if anyone out there has any solution to this problem. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dkwok Sent: Friday, February 20, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: codec negotiation prob solved (Philipp von Klitzing) wrote: FYI - bug 1043 has been fixed on Feb 18: "From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec. In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a "Codec Match". Asterisk needs to do a bit compare, and not a int compare in this case." -- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 >> I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out. My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition [general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=yes dropcount=3 maxjitterbuffer=250 maxexccessbuffer=50 register => dkwok:[EMAIL PROTECTED] tos=lowdelay [iax_home] type=friend context=int-ext auth=md5 user=iax_home secret=cc trunking=yes disallow=all allow=gsm host=dynamic qualify=yes [iaxtel] type=friend disallow=all disallow=g729 allow=gsm trunking=yes context=from-iaxtel [atp] type=friend disallow=all allow=g729 trunking=yes context=atp host=xxx.xxx.xxx.xxx I would like to hear any comment from * developer. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 troubles
I too had lot of issues with G729 license, I too was asked to start asterisk with a console option. Without it asterisk stop running. I could never do a safe_asterisk start with the g729 license installed. Recently I rebooted my server and now I can start asterisk with safe_asterisk. So Reboot of the server would have done something. This may be useful if you are having license issues with the G729 codecs. Cheers! -SamW -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Monday, March 01, 2004 12:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles Thanks Wes, I just tried it but it does not seem to make any difference. Darren Wiebe Wes Marderness wrote: >My server is running fine now. I have to 'cd /tmp' then >'/usr/sbin/asterisk -gc' or I receive error messages. It is very strange >but it works. > >Wes > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe >Sent: Monday, March 01, 2004 11:23 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] G729 troubles > > >I have gotten in contact with Digium. As soon as we have something >resolved, I will post the fix to the list. > >Thanks, >Darren Wiebe >[EMAIL PROTECTED] > >Brent Franks wrote: > > > >>Wes, >> >>Please let us know how you make out with this. >> >>I experience the same issues. >> >>- Brent >> >> >> >> >> >>>-Original Message- >>>From: [EMAIL PROTECTED] [mailto:asterisk-users- >>>[EMAIL PROTECTED] On Behalf Of Wes Marderness >>>Sent: Monday, March 01, 2004 9:24 AM >>>To: [EMAIL PROTECTED] >>>Subject: RE: [Asterisk-Users] G729 troubles >>> >>>I had a problem something like what you described. I was able to run >>>asterisk from /tmp, had to bug digium for an answer for that one. Hope >>>this >>>helps >>> >>>wes >>> >>>-Original Message- >>>From: [EMAIL PROTECTED] >>>[mailto:[EMAIL PROTECTED] Behalf Of Darren >>> >>> >>> >>> >>Wiebe >> >> >> >> >>>Sent: Saturday, February 28, 2004 2:10 PM >>>To: [EMAIL PROTECTED] >>>Subject: Re: [Asterisk-Users] G729 troubles >>> >>> >>>I forgot to mention what I have been trying to fix it. I'm running it >>> >>> >>>from the console "asterisk -vvvcng" but this does not help. I've >> >> >>>searched the mailing lists and found a lot of messages with people >>>having the same problem. I'll try calling digium Monday if I cannot >>>resolve it today and see if they can help me. >>>Darren Wiebe >>>[EMAIL PROTECTED] >>> >>>Darren Wiebe wrote: >>> >>> >>> >>> >>> I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm >>getting >> >> >> >> from Asterisk. Can anybody help me? >>> >>> [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: >>Select >> >> >> >> retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: >>Select >> >> >> >> retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>>___ >>>Asterisk-Users mailing list >>>[EMAIL PROTECTED] >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>>___ >>>Asterisk-Users mailing list >>>[EMAIL PROTECTED] >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> > >__
[Asterisk-Users] Supervised transfer (almost) with GS phone
I have now tested a (previously suggested) method for doing supervised transfers using the Grandstream SIP phone. It isn't perfect, but it works and is very functional. Here are the steps: 1) A call comes in that you want to transfer 2) Flash once. This switches to the 2nd line and puts the caller on hold. 3) Call the party to which you want to transfer the call. 4) Arrange the transfer and have the person to which the call is being transferred hang up. Incidentally, while it may be obvious, it is still worth mentioning that before they hang up, you can switch back and forth between the two parties an indefinate number of times. 5) WITHOUT HANGING UP, press the flash button again. This switches you back to the caller. 6) Inform them of the transfer. 7) Press the transfer button and enter the transfer phone number. (This starts a simple blind transfer). 8) Hang up to complete the blind transfer. Relative to a true supervised transfer, the only things I believe to be missing are the inability to transfer the call while the desired recipient is on the line, and the ability to pick the call back up if the transfer fails - and these may be major problems for some applications. However, it comes close enough for many purposes. For reference, I'm using a slightly modified version of the stdexten macro in the dialplan (no t or T) and either SIP INFO or INBAND on the phones. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not ringing.
Mark Messmore, Technical Support, University Telcom Inc. wrote: > I was just wondering if anyone has had this situation...or one > similar to it. > > I've got a Sipura SPA 2000. After hooking it up and configuring it > with my * box, it has worked well. From both lines we are able to > dial out at any point in time. However after a few minutes (5-10 > usually) the Sipura will stop sending a ringer signal to the phones. > We can still dial, * shows that it is ringing those SIP clients, and > both lines are still shown as being "registered". Everything will > work fine for the first 5-10 minutes after being "rebooted"...however > after that there is silence. Even if I dial the line from my cell > phone and pick up the telephone that I am dialing...there is > dial-tone...therefore it seems that the Sipura is just not passing > those incoming calls after a short time period. > > If you have any idea why this is happening I'd sure appreciate > hearing your thoughts. Thanks > I had (and still have) similar problem. Once SPA 2000 registers with * it all works well for few minutes. After that all incoming calls are not answered by SPA 2000. Is that what you mean? If so, I have temporaraly got SPA 2000 to re-register every 3 minutes. This seems to work at the moment. (Sipura tech support known about this problem for about 6-8 weeks now, but I am still to hear from them). Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not ringing.
Well thanks for letting me know that I'm not the only one having this problem. I'll be sure to try that out...that should at least be a temporary fix. I've also put in a tech-support ticket through Atacom...who I got the SPA 2000 from. I hope that we can get this all figured out. If I hear anything I'll be sure to let you know. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Tuesday, March 02, 2004 3:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura 2000 not ringing. Mark Messmore, Technical Support, University Telcom Inc. wrote: > I was just wondering if anyone has had this situation...or one > similar to it. > > I've got a Sipura SPA 2000. After hooking it up and configuring it > with my * box, it has worked well. From both lines we are able to > dial out at any point in time. However after a few minutes (5-10 > usually) the Sipura will stop sending a ringer signal to the phones. > We can still dial, * shows that it is ringing those SIP clients, and > both lines are still shown as being "registered". Everything will > work fine for the first 5-10 minutes after being "rebooted"...however > after that there is silence. Even if I dial the line from my cell > phone and pick up the telephone that I am dialing...there is > dial-tone...therefore it seems that the Sipura is just not passing > those incoming calls after a short time period. > > If you have any idea why this is happening I'd sure appreciate > hearing your thoughts. Thanks > I had (and still have) similar problem. Once SPA 2000 registers with * it all works well for few minutes. After that all incoming calls are not answered by SPA 2000. Is that what you mean? If so, I have temporaraly got SPA 2000 to re-register every 3 minutes. This seems to work at the moment. (Sipura tech support known about this problem for about 6-8 weeks now, but I am still to hear from them). Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SPAM[RBL] RE: [Asterisk-Users] G729 troubles
I don't know if all these issues have the same root problem. John Bigelow from digium was connected to my system this morning. He was going to fix something in the source. It is supposed to be working for me in a few hours. Darren Wiebe [EMAIL PROTECTED] SamW wrote: I too had lot of issues with G729 license, I too was asked to start asterisk with a console option. Without it asterisk stop running. I could never do a safe_asterisk start with the g729 license installed. Recently I rebooted my server and now I can start asterisk with safe_asterisk. So Reboot of the server would have done something. This may be useful if you are having license issues with the G729 codecs. Cheers! -SamW -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Monday, March 01, 2004 12:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles Thanks Wes, I just tried it but it does not seem to make any difference. Darren Wiebe Wes Marderness wrote: My server is running fine now. I have to 'cd /tmp' then '/usr/sbin/asterisk -gc' or I receive error messages. It is very strange but it works. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe Sent: Monday, March 01, 2004 11:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles I have gotten in contact with Digium. As soon as we have something resolved, I will post the fix to the list. Thanks, Darren Wiebe [EMAIL PROTECTED] Brent Franks wrote: Wes, Please let us know how you make out with this. I experience the same issues. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wes Marderness Sent: Monday, March 01, 2004 9:24 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G729 troubles I had a problem something like what you described. I was able to run asterisk from /tmp, had to bug digium for an answer for that one. Hope this helps wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe Sent: Saturday, February 28, 2004 2:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles I forgot to mention what I have been trying to fix it. I'm running it from the console "asterisk -vvvcng" but this does not help. I've searched the mailing lists and found a lot of messages with people having the same problem. I'll try calling digium Monday if I cannot resolve it today and see if they can help me. Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/li
RE: [Asterisk-Users] RE: codec negotiation prob solved
Dear All, That is my experience with Asterisk too, this codec negotiation is giving us lots of problems. I am using Asterisk mostly for passing-through VOIP traffic. Basically I will have to choose g7231 or g729 and not both. If I choose both, and when calls come in with with both codecs, and the terminating gateway (endpoint) only allows g729, the calls would go through, but if the terminating gateway only allows g7231, then calls would not go through, and if the terminating gateway allows both codecs, call would not go through either. Worst yet, Asterisk seems not to work with t38 fax and I have to allow g711 in order to get fax to go through and that is ONLY between Asterisk, if a cisco calls into my Asterisk with a fax, it just will not work. Anyway, the worst part is if I allow g711 on my Asterisk, ALL calls coming into my Asterisk will get converted to g711 before going out, whether out to a third party equipment (if they allow g711) or my other Asterisk. Conversion takes up too much resource and needless to say bandwidth (for g711), and it is highly a NONO to convert to g711 for all calls, even voice calls. If I allow g723, g729 and g711, calls will NOT just get passthrough, they will get converted. Is there anyway we can debug this problem please? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of SamW Sent: Tuesday, March 02, 2004 3:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved I agree, that * codec negotiation is buggy, there must be some mechanism to give priority to pass through without trying to codec translate. Codec translation need lot of CPU and can deteriorate quality by its nature. Some developer sheding some light on this is buggy codec translation is very appreciated. - SamW -Original Message- From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Friday, February 20, 2004 12:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved I have the same problem, most carriers out there deal with both g723.1 or g729. During passing through via Asterisk, carrier customers will send us calls broadcasting both codecs with one having priority over the other, the way it is supposed to work is that asterisk will try to negotiate the top priority codec first with the terminating endpoint, assuming that the originating endpoint broadcasts g729 as first priority and then g723.1, Asterisk should take g729 and try to negotiate with terminating endpoint and if the terminating takes g729, then the call should be patched and bridged, but if the terminating endpoint takes ONLY g723.1, then Asterisk should then go back and take g723.1 (which is the second priority as per the originating endpoint) and bridge the call through. However, the way Asterisk is doing it is if I allow both g723.1 and g729, then if the originating endpoint broadcasts both codecs and the terminating endpoint only allows g723.1, then the call will not go through and it will say no path from g729 to ., and calls will not go through. Summing up, if originating gateway allows both g723.1 and g729 , Asterisk being the pass-through entity, allows both codecs, and the terminating gateway allows ONLY g723.1, the calls will not go through which is certainly a bug in the asterisk. I wonder if anyone out there has any solution to this problem. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dkwok Sent: Friday, February 20, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: codec negotiation prob solved (Philipp von Klitzing) wrote: FYI - bug 1043 has been fixed on Feb 18: "From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec. In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a "Codec Match". Asterisk needs to do a bit compare, and not a int compare in this case." -- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 >> I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out. My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition [general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10
[Asterisk-Users] H323 calls drop on connect
I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 not ringing.
> Mark Messmore, Technical Support, University Telcom Inc. wrote: > > I was just wondering if anyone has had this situation...or one > > similar to it. > > > > I've got a Sipura SPA 2000. After hooking it up and configuring it > > with my * box, it has worked well. From both lines we are able to > > dial out at any point in time. However after a few minutes (5-10 > > usually) the Sipura will stop sending a ringer signal to the phones. > > We can still dial, * shows that it is ringing those SIP clients, and > > both lines are still shown as being "registered". Everything will > > work fine for the first 5-10 minutes after being "rebooted"...however > > after that there is silence. Even if I dial the line from my cell > > phone and pick up the telephone that I am dialing...there is > > dial-tone...therefore it seems that the Sipura is just not passing > > those incoming calls after a short time period. > > > > If you have any idea why this is happening I'd sure appreciate > > hearing your thoughts. Thanks > > > > I had (and still have) similar problem. Once SPA 2000 registers with * > it all works well for few minutes. After that all incoming calls are not > answered by SPA 2000. > Is that what you mean? > > If so, I have temporaraly got SPA 2000 to re-register every 3 minutes. > This seems to work at the moment. Are you sure you configured the SPA to send a keep alive message every 15-30 seconds? Sounds to me like the NAT binding is simply closing up and getting reopened with the frequent registrations. > (Sipura tech support known about this problem for about 6-8 weeks now, > but I am still to hear from them). > > Ta > SJ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 fax (off-topic)
Does somebody now if there is some opensource software which can handle T.38 SIP and convert it to a tiff or something ? Kind regards Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 calls drop on connect
Hi, Todd Did you notice that when you made the calls, were the calls indicated as "answered", in both cases? And if so, did the indication "answered" pop up when the calls were actually picked up and answered or right after the call setup was completed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr->dst incorrect? Bug submitted.
Bug submitted, for this missing functionality. Bug ID : 1141. Thanks for who ever contributed. - SamW -Original Message- From: Philipp von Klitzing [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 25, 2004 2:45 PM To: SamW Subject: RE: [Asterisk-Users] cdr->dst incorrect? Hi! > not work correctly. All what I did recently was upgrading asterisk to > latest version. Can this be a bug? I would say so and suggest that you report this at bugs.digium.com > Can this be reproduced else where. (I haven't seen any complains on the > message board other than me.) Yes - the answer of OEJ was a reaction to the same problem that I posted on this list. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax (off-topic)
Michael, Check out T38Modem at www.openh323.org -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: "Michael Devenijn" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, March 02, 2004 4:43 PM Subject: [Asterisk-Users] T.38 fax (off-topic) Does somebody now if there is some opensource software which can handle T.38 SIP and convert it to a tiff or something ? Kind regards Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS "TX" record?
John Todd wrote: Are you aware of the +878 "country" code and the UPT (Universal Personal Telecommunications) project? Yes I am aware of that range, and I'm also aware we may never be allocated the range either, but if we don't try these things and pack up and go home we will never know either. However how hard/easy will it be to add tel fields, or other enum information with visionng.org ? -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax (off-topic)
On Tue, 2 Mar 2004, Darren Nickerson wrote: > Check out T38Modem at www.openh323.org Is there any way to actually get that working with Asterisk yet? As far as I can tell, neither of the H323 implementations for Asterisk support the T.38 protocol. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VTGO-PG and IPP200
Tim, A bit more research on the Wiki pages and I found these links: chan_sccp An alternative to chan_skinny. http://www.zozo.org.uk/pages.shtml?page=sccp (cache) (Original) http://www.lambda-solutions.de/7920/ (cache) (Spin-Off with CVS & Mantis) Perhaps the chan_sccp is further along the the chan_skinny. I guess we'll just have to try them both. Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Tuesday, March 02, 2004 10:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VTGO-PG and IPP200 On Tue, Mar 02, 2004 at 10:40:24AM -0800, Ed Rubright wrote: > The VTGO-PC appears to be Cisco Call Manager only or did you find > somewhere on there site that they have a SIP version? I'm bugging them about the SIP, but so far, no answer. They claim it's a software close of the Cisco hardphone, so... > If it is Call Manager only...how far along is Asterisk support for > Call Manager? I looked on the Wiki pages and saw mentioned the 2 > implementations of the "skinny" protocol, but nothing about its current feature state. I'm not sure what state it's in. Or how to configure it for that matter. I may grab the eval copy and try to crash * with it... :) Tim -- ><<< ><< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << ><<< ><< ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax (off-topic)
On Tue, 2 Mar 2004, Nate Carlson wrote: > On Tue, 2 Mar 2004, Darren Nickerson wrote: > > Check out T38Modem at www.openh323.org > > Is there any way to actually get that working with Asterisk yet? > As far as I can tell, neither of the H323 implementations for Asterisk > support the T.38 protocol. Last time I tried that with openh323 it didn't work either. It would get a page in or out and then crash and burn. -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Phones
We have worked with some of you supplying Cisco IP phones to use with your Asterisk system. We have a good supply of new and used Cisco IP phones (CP-7905, CP-7940 and CP-7960) at this time. All of our equipment carries a 90-day warranty. If we can help you out with any of your phone, router or switch needs, please let me know. Thanks, Jon PutnamGlobal Technology Solutionsmain #763-488-1870 #222fax #763-488-1875[EMAIL PROTECTED]AIM: GTSJonPwww.gtsinc.biz
