RE: [Asterisk-Users] SIP Listen to multiple ports
You must have missed the part where Prepaid got upset when I suggested workarounds :) > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. > Sent: Tuesday, June 14, 2005 12:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > Hi, > > What about using IPTABLES DNAT stuff in order to map all incoming > 5061 traffic to 5060 port ? That may work. > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > Conclusion - asterisk only listens on one port, and ignores the second > > port= or bindport= > > -- > Juanjo sin .sig :( > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
If your app is .net get the .net provider for mysql and give it to your dba/programmer with the docs, he/she will figure it out. No different than talking to ms sql with .net except you reference a different data provider. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Tuesday, June 14, 2005 12:43 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on in DB > > Thanks for info . How do you integrate * specific data in mysql with data > from MSSQL? App is running on .NET, in this case it will need to have > assess to both DBs and update them simultaneously. Sorry, I'm not a DB > admin. > > > I.N. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep > Sent: Monday, June 13, 2005 11:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on > in DB > > Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL > stink Odbc in general stinks. > > You might be able to get MS SQL DTS (data transformation services) to > link to the mysql database and present the data as it were in your ms > sql database. > > There is a pretty good odbc 3.51 mysql driver for windows, as well as a > .net provider. Both at www.mysql.org. > > Mysql is free, * will talk to is using the native TDS > > You can run the windows version of mysql on a windows box if you wish, > but why? Faster if it is on the same box as asterisk unless * is heavily > loaded. > > I tried the * > realtime > odbc > mssql thing, gave up after having poor > results getting the various ms sql drivers for linux to work right. > > our main app uses data in ms sql and mysql and there is a common key in > the data to link accounting data with the * user data for views where > they are both required. > > We also use mysql for cdr for billing purposes. > > I was much more comfortable with .net ms sql, but the transition and > integration with mysql was easy. Just store the asterisk specific data > in mysql, everthing else in ms sql if you must. > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > > Sent: Tuesday, June 14, 2005 12:04 AM > > To: 'Shamsul Arefin'; 'Asterisk Users Mailing List - Non-Commercial > > Discussion' > > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and > so > > on in DB > > > > Could you go with some details? What was better performance, > stability? > > All our user info is in MS SQL and we have billing based on it, so it > > won't > > be easy to move to mysql. > > > > I.N. > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul > > Arefin > > Sent: Monday, June 13, 2005 10:36 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and > so > > on > > in DB > > > > Yes it is now possible to store configuration files in database, via > > Mysql support or via ODBC. But we have find that Mysql is works much > > better. > > > > regards > > shams > > > > On 6/14/05, Irakli Natsvlishvili <[EMAIL PROTECTED]> wrote: > > > Hello, > > > > > > I have one question regarding *. Default configuration for asterisk > is > > to > > > keep configuration(s) in ordinary text based config files. > > > > > > My question is simple: is it possible to keep those config info (at > > least, > > > to start from - sip.conf, extensions.conf and voicemail.conf) on a > > database, > > > which asterisk access via ODBC. If it is possible, I'd appreciate if > > someone > > > points me where I can read more about it and shows me some examples. > > Also > > > I'd like to know, how asterisk behaves (in terms of stability and > > > performance) in this environment. > > > > > > > > > I.N. > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > Best Regards > > Shamsul Arefin > > Saktek Technologies > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > 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 > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >
Re: [Asterisk-Users] SIP Listen to multiple ports
Hi, What about using IPTABLES DNAT stuff in order to map all incoming 5061 traffic to 5060 port ? That may work. On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > Conclusion - asterisk only listens on one port, and ignores the second > port= or bindport= -- Juanjo sin .sig :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Thanks for info . How do you integrate * specific data in mysql with data from MSSQL? App is running on .NET, in this case it will need to have assess to both DBs and update them simultaneously. Sorry, I'm not a DB admin. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL stink Odbc in general stinks. You might be able to get MS SQL DTS (data transformation services) to link to the mysql database and present the data as it were in your ms sql database. There is a pretty good odbc 3.51 mysql driver for windows, as well as a .net provider. Both at www.mysql.org. Mysql is free, * will talk to is using the native TDS You can run the windows version of mysql on a windows box if you wish, but why? Faster if it is on the same box as asterisk unless * is heavily loaded. I tried the * > realtime > odbc > mssql thing, gave up after having poor results getting the various ms sql drivers for linux to work right. our main app uses data in ms sql and mysql and there is a common key in the data to link accounting data with the * user data for views where they are both required. We also use mysql for cdr for billing purposes. I was much more comfortable with .net ms sql, but the transition and integration with mysql was easy. Just store the asterisk specific data in mysql, everthing else in ms sql if you must. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Tuesday, June 14, 2005 12:04 AM > To: 'Shamsul Arefin'; 'Asterisk Users Mailing List - Non-Commercial > Discussion' > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on in DB > > Could you go with some details? What was better performance, stability? > All our user info is in MS SQL and we have billing based on it, so it > won't > be easy to move to mysql. > > I.N. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul > Arefin > Sent: Monday, June 13, 2005 10:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and so > on > in DB > > Yes it is now possible to store configuration files in database, via > Mysql support or via ODBC. But we have find that Mysql is works much > better. > > regards > shams > > On 6/14/05, Irakli Natsvlishvili <[EMAIL PROTECTED]> wrote: > > Hello, > > > > I have one question regarding *. Default configuration for asterisk is > to > > keep configuration(s) in ordinary text based config files. > > > > My question is simple: is it possible to keep those config info (at > least, > > to start from - sip.conf, extensions.conf and voicemail.conf) on a > database, > > which asterisk access via ODBC. If it is possible, I'd appreciate if > someone > > points me where I can read more about it and shows me some examples. > Also > > I'd like to know, how asterisk behaves (in terms of stability and > > performance) in this environment. > > > > > > I.N. > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Best Regards > Shamsul Arefin > Saktek Technologies > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
I just ran a couple of test with CVS Head Port=5060 Port=5061 Result = chan_sip reports listening on 5060 Port=5061 Port=5060 Result = chan_sip reports listening on 5060 (ignoring port=?) Port=5061 Result = chan_sip STILL reports listening on 5060 Bindport=5061 Result = chan_sip reports listening on 5061 Bindport=5061 Bindport=5060 Result = istening on 5060 Conclusion - asterisk only listens on one port, and ignores the second port= or bindport= > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Monday, June 13, 2005 11:41 PM > To: Prepaid; Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > Hello > > I added to sip.conf 2 instances of the port parameter ej: > > [general] > port=5060 > port=5061 > > It works for me, we used the Linksys pap2-na with both lines at the same > time > we cant bind it to the same 5060 port, then I configured line1 to 5060 and > line2 > to 5061, asterisk is listening at both ports and working like expected > > Juan Bou > > > At 12:09 a.m. 14/06/2005, you wrote: > > > > > > > > > > -Original Message- > > > > > > > > From: [EMAIL PROTECTED] > > > > > [mailto:asterisk-users- > > > > > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > > > > > To: asterisk-users@lists.digium.com > > > > > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > > > > > > > Hello all > > > > > > > > > > > > > > > > I'm trying to get my asterisk config to listen to multiple > > > ports. > > > > > This > > > > > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > > > > > supported? > > > > > > > > > > > > > > > > Thanks! > > > > > > > > ___ > > > > > > > > Asterisk-Users mailing list > > > > > > > > Asterisk-Users@lists.digium.com > > > > > > > > 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 > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ~ > > - > > Juan Bou Riquer. > > Internet Cancun. > > [EMAIL PROTECTED] > > Tel. 87-2601 Fax. 84-3809 > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL stink Odbc in general stinks. You might be able to get MS SQL DTS (data transformation services) to link to the mysql database and present the data as it were in your ms sql database. There is a pretty good odbc 3.51 mysql driver for windows, as well as a .net provider. Both at www.mysql.org. Mysql is free, * will talk to is using the native TDS You can run the windows version of mysql on a windows box if you wish, but why? Faster if it is on the same box as asterisk unless * is heavily loaded. I tried the * > realtime > odbc > mssql thing, gave up after having poor results getting the various ms sql drivers for linux to work right. our main app uses data in ms sql and mysql and there is a common key in the data to link accounting data with the * user data for views where they are both required. We also use mysql for cdr for billing purposes. I was much more comfortable with .net ms sql, but the transition and integration with mysql was easy. Just store the asterisk specific data in mysql, everthing else in ms sql if you must. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Tuesday, June 14, 2005 12:04 AM > To: 'Shamsul Arefin'; 'Asterisk Users Mailing List - Non-Commercial > Discussion' > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on in DB > > Could you go with some details? What was better performance, stability? > All our user info is in MS SQL and we have billing based on it, so it > won't > be easy to move to mysql. > > I.N. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul > Arefin > Sent: Monday, June 13, 2005 10:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and so > on > in DB > > Yes it is now possible to store configuration files in database, via > Mysql support or via ODBC. But we have find that Mysql is works much > better. > > regards > shams > > On 6/14/05, Irakli Natsvlishvili <[EMAIL PROTECTED]> wrote: > > Hello, > > > > I have one question regarding *. Default configuration for asterisk is > to > > keep configuration(s) in ordinary text based config files. > > > > My question is simple: is it possible to keep those config info (at > least, > > to start from - sip.conf, extensions.conf and voicemail.conf) on a > database, > > which asterisk access via ODBC. If it is possible, I'd appreciate if > someone > > points me where I can read more about it and shows me some examples. > Also > > I'd like to know, how asterisk behaves (in terms of stability and > > performance) in this environment. > > > > > > I.N. > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Best Regards > Shamsul Arefin > Saktek Technologies > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
If you do a sip show peers I think you will see that your PAP2 setup registers its port with * as being 5060 on line 1 and 5061 on line 2, but it stills calls port 5060 on asterisk when it makes the registration. I think * is actually listening on the first configured port. You might get the same results you have now after removing the port=5061, have you tried Prepaid, post if this actually works or not in your case where port 5060 out from the UA is blocked. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Monday, June 13, 2005 11:41 PM > To: Prepaid; Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > Hello > > I added to sip.conf 2 instances of the port parameter ej: > > [general] > port=5060 > port=5061 > > It works for me, we used the Linksys pap2-na with both lines at the same > time > we cant bind it to the same 5060 port, then I configured line1 to 5060 and > line2 > to 5061, asterisk is listening at both ports and working like expected > > Juan Bou > > > At 12:09 a.m. 14/06/2005, you wrote: > > > > > > > > > > -Original Message- > > > > > > > > From: [EMAIL PROTECTED] > > > > > [mailto:asterisk-users- > > > > > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > > > > > To: asterisk-users@lists.digium.com > > > > > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > > > > > > > Hello all > > > > > > > > > > > > > > > > I'm trying to get my asterisk config to listen to multiple > > > ports. > > > > > This > > > > > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > > > > > supported? > > > > > > > > > > > > > > > > Thanks! > > > > > > > > ___ > > > > > > > > Asterisk-Users mailing list > > > > > > > > Asterisk-Users@lists.digium.com > > > > > > > > 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 > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ~ > > - > > Juan Bou Riquer. > > Internet Cancun. > > [EMAIL PROTECTED] > > Tel. 87-2601 Fax. 84-3809 > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] problem with pf and asterisk
On 20:00, Mon 13 Jun 05, Frank Cases wrote: > current setup > > SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- > (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk > > > problem is RTP stream not oging trouhg from * to sip and vice versa. > > #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as > return address > or > #2 asterisk trying to get back to me as 192.168 on public internet.. > > > > got > canreinvite=yes and no. > nat=yes > qualify=1000 > > externaladdr=IP of (em1) > localnet=172.16.0.0/12 > > Try to set the externaladdr to the IP of xl0. That did the trick for me here. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Could you go with some details? What was better performance, stability? All our user info is in MS SQL and we have billing based on it, so it won't be easy to move to mysql. I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shamsul Arefin Sent: Monday, June 13, 2005 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB Yes it is now possible to store configuration files in database, via Mysql support or via ODBC. But we have find that Mysql is works much better. regards shams On 6/14/05, Irakli Natsvlishvili <[EMAIL PROTECTED]> wrote: > Hello, > > I have one question regarding *. Default configuration for asterisk is to > keep configuration(s) in ordinary text based config files. > > My question is simple: is it possible to keep those config info (at least, > to start from - sip.conf, extensions.conf and voicemail.conf) on a database, > which asterisk access via ODBC. If it is possible, I'd appreciate if someone > points me where I can read more about it and shows me some examples. Also > I'd like to know, how asterisk behaves (in terms of stability and > performance) in this environment. > > > I.N. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Shamsul Arefin Saktek Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] SIP-H.323 dial tone and busy tone problem.
Try playing with faststart . Moises Silva wrote: Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made? best regards On 6/11/05, Carlos Alberto Lara de Hoyos <[EMAIL PROTECTED]> wrote: Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. SIP:[EMAIL PROTECTED] Tel:(+359-2) 856-19-43 400-55-65 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 3, fl. 1, entr. A, "Preki pat" str., No. 16, kv. Pavlovo, 1618 Sofia, Bulgaria ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 connecting remote villages in western Uganda
Mark, What you are doing is indeed wonderful and I applaud you for your efforts! I will be sending in my details in an email soon. By the way if you (or any other non profit organisation for the matter) are interested in similar efforts going on in Asia do drop me an email as I am based in Singapore. There is also a great need for rebuilding in many parts such as the Tsunami-hit areas, etc. Keep up the good work, and as some say "May the force be with you!" :) Cheers Caleb On 6/14/05, Mark Summer <[EMAIL PROTECTED]> wrote: > Hi, > > I though some of you on this list might be interested in what Inveneo > is doing in Uganda. We are a San Francisco based non-profit > organization that builds rugged, low-cost, highly reliable and open- > source communications systems for under-served communities around the > world. We have just completed our first installation in western > Uganda, Africa. > > The system is up and running since this past Wednesday (June 8th). We > have installed 5 units, 4 of which are in villages with with no > access to power. The system provides Internet access and phone > capabilities to the users. Phone calls among the connected villages > are free of charge, with the ability to place and receive calls to / > from the Ugandan phone network and voice mail boxes for each > station. The systems are linked using 802.11 WiFi links. > > For more information please have a look at the following links: > > For more detailed information and pictures of the Uganda deployment: > > http://www.inveneo.org/?q=uganda > > For more information about the solution we have built and > implemented, here is a link to our PDF datasheet: > > http://www.inveneo.org/download/inveneoDatasheet.pdf > > And of course our website: > > http://www.inveneo.org/ > > > Thank you! > > Mark > > > Mark Summer > co-founder, Inveneo > web: http://www.inveneo.org > phone: +1-415-901-1969 x 1200 > FWD: 603303 > cell: +1-415-867-9751 > email: [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Not as fallback, but both can be used together; You can have some static info in the test files and some in realtime, * will use the sum of both. The main benefit to RT is the reduction in the need to "reload" Read the wiki, these answers and much more are in there. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Monday, June 13, 2005 11:40 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on in DB > > When in realtime mode, does * uses static configs at all? Is it possible > to > operate in realtime mode and have static configs (which are build based on > information taken from DB) as fallback solution? > > > I.N. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep > Sent: Monday, June 13, 2005 10:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so > on > in DB > > As far as performance, * caches static config, but queries realtime > configs, so scalability must be impacted, but I personally have not > approached the limits of realtime yet. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adtran TA 750 FXO Groundstart Mode
I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
Hello I added to sip.conf 2 instances of the port parameter ej: [general] port=5060 port=5061 It works for me, we used the Linksys pap2-na with both lines at the same time we cant bind it to the same 5060 port, then I configured line1 to 5060 and line2 to 5061, asterisk is listening at both ports and working like expected Juan Bou At 12:09 a.m. 14/06/2005, you wrote: > > > > > > -Original Message- > > > > > > From: [EMAIL PROTECTED] > > > [mailto:asterisk-users- > > > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > > > To: asterisk-users@lists.digium.com > > > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > > > Hello all > > > > > > > > > > > > I'm trying to get my asterisk config to listen to multiple > ports. > > > This > > > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > > > supported? > > > > > > > > > > > > Thanks! > > > > > > ___ > > > > > > Asterisk-Users mailing list > > > > > > Asterisk-Users@lists.digium.com > > > > > > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ~ - Juan Bou Riquer. Internet Cancun. [EMAIL PROTECTED] Tel. 87-2601 Fax. 84-3809 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
When in realtime mode, does * uses static configs at all? Is it possible to operate in realtime mode and have static configs (which are build based on information taken from DB) as fallback solution? I.N. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Keeping users, extensions,voicemail and so on in DB As far as performance, * caches static config, but queries realtime configs, so scalability must be impacted, but I personally have not approached the limits of realtime yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
For Asterisk to work properly, you will have to bind to one port only. Regards Shams On 6/14/05, Prepaid <[EMAIL PROTECTED]> wrote: > Yes they could, then I would just switch to VoIP or VPN and it > wouldn't be possible for them to do that, anything is possible after > all. > > Please help me find a solution to my original request instead of > trying to go all the way around it. > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > You don't have to block ports 10k to 20k to block rtp, they could block > > packets matching the RTP profile, under 100k in size, udp, port above > > 10k, 20ms timing... > > > > > -Original Message- > > > From: Prepaid [mailto:[EMAIL PROTECTED] > > > Sent: Monday, June 13, 2005 11:04 PM > > > To: Damon Estep; asterisk-users@lists.digium.com > > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > > > The RTP ports are not blocked, I have already binded on the other port > > > and it works fine, I would just like to have Asterisk bind to multiple > > > ports. > > > > > > If they start blocking the 10,000 to 20,000 ports for RTP it will > > > never end and they might as well not allow connections on any out > > > going ports besides 80. > > > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > > > > > > You will not win, once you get past the 5060 issue you might find > > RTP > > > > blocked also. > > > > > > > > > > > > > -Original Message- > > > > > From: Prepaid [mailto:[EMAIL PROTECTED] > > > > > Sent: Monday, June 13, 2005 10:54 PM > > > > > To: Damon Estep; asterisk-users@lists.digium.com > > > > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > The end point that is having the issue is not in the US. > > > > > > > > > > I could just change the default port on asterisk to another one, > > but I > > > > > would also like it to bind in 5060 in addition to the extra ports > > for > > > > > "special isps" > > > > > > > > > > The ISP blocking the port 5060 is Airtel in Bangalore, India, for > > the > > > > > curiosity of others.. > > > > > > > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > > > > Are you in the USA? > > > > > > > > > > > > If so call the FCC, they do not like port 5060 blocking (or any > > > > other > > > > > > VoIP port blocking) > > > > > > > > > > > > See here: http://www.google.com/search?hl=en&q=fcc+fine+voip > > > > > > > > > > > > Not the technical answer you are looking for but the RIGHT > > answer. > > > > > > > > > > > > > -Original Message- > > > > > > > From: [EMAIL PROTECTED] > > > > [mailto:asterisk-users- > > > > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > > > > To: asterisk-users@lists.digium.com > > > > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > > > > > Hello all > > > > > > > > > > > > > > I'm trying to get my asterisk config to listen to multiple > > ports. > > > > This > > > > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > > > > supported? > > > > > > > > > > > > > > Thanks! > > > > > > > ___ > > > > > > > Asterisk-Users mailing list > > > > > > > Asterisk-Users@lists.digium.com > > > > > > > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Shamsul Arefin Saktek Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Yes it is now possible to store configuration files in database, via Mysql support or via ODBC. But we have find that Mysql is works much better. regards shams On 6/14/05, Irakli Natsvlishvili <[EMAIL PROTECTED]> wrote: > Hello, > > I have one question regarding *. Default configuration for asterisk is to > keep configuration(s) in ordinary text based config files. > > My question is simple: is it possible to keep those config info (at least, > to start from - sip.conf, extensions.conf and voicemail.conf) on a database, > which asterisk access via ODBC. If it is possible, I'd appreciate if someone > points me where I can read more about it and shows me some examples. Also > I'd like to know, how asterisk behaves (in terms of stability and > performance) in this environment. > > > I.N. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Shamsul Arefin Saktek Technologies ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
As far as performance, * caches static config, but queries realtime configs, so scalability must be impacted, but I personally have not approached the limits of realtime yet. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Monday, June 13, 2005 11:22 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Keeping users, extensions,voicemail and so on in > DB > > Hello, > > I have one question regarding *. Default configuration for asterisk is to > keep configuration(s) in ordinary text based config files. > > My question is simple: is it possible to keep those config info (at least, > to start from - sip.conf, extensions.conf and voicemail.conf) on a > database, > which asterisk access via ODBC. If it is possible, I'd appreciate if > someone > points me where I can read more about it and shows me some examples. Also > I'd like to know, how asterisk behaves (in terms of stability and > performance) in this environment. > > > I.N. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] Keeping users, extensions, voicemail and so on in DB
Search for "asterisk realtime" at www.voip-info.org Answer is yes, mysql or odbc. Requires head, not stable. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili > Sent: Monday, June 13, 2005 11:22 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Keeping users, extensions,voicemail and so on in > DB > > Hello, > > I have one question regarding *. Default configuration for asterisk is to > keep configuration(s) in ordinary text based config files. > > My question is simple: is it possible to keep those config info (at least, > to start from - sip.conf, extensions.conf and voicemail.conf) on a > database, > which asterisk access via ODBC. If it is possible, I'd appreciate if > someone > points me where I can read more about it and shows me some examples. Also > I'd like to know, how asterisk behaves (in terms of stability and > performance) in this environment. > > > I.N. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Keeping users, extensions, voicemail and so on in DB
Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can read more about it and shows me some examples. Also I'd like to know, how asterisk behaves (in terms of stability and performance) in this environment. I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
Yes they could, then I would just switch to VoIP or VPN and it wouldn't be possible for them to do that, anything is possible after all. Please help me find a solution to my original request instead of trying to go all the way around it. On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > You don't have to block ports 10k to 20k to block rtp, they could block > packets matching the RTP profile, under 100k in size, udp, port above > 10k, 20ms timing... > > > -Original Message- > > From: Prepaid [mailto:[EMAIL PROTECTED] > > Sent: Monday, June 13, 2005 11:04 PM > > To: Damon Estep; asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > The RTP ports are not blocked, I have already binded on the other port > > and it works fine, I would just like to have Asterisk bind to multiple > > ports. > > > > If they start blocking the 10,000 to 20,000 ports for RTP it will > > never end and they might as well not allow connections on any out > > going ports besides 80. > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > > > > You will not win, once you get past the 5060 issue you might find > RTP > > > blocked also. > > > > > > > > > > -Original Message- > > > > From: Prepaid [mailto:[EMAIL PROTECTED] > > > > Sent: Monday, June 13, 2005 10:54 PM > > > > To: Damon Estep; asterisk-users@lists.digium.com > > > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > The end point that is having the issue is not in the US. > > > > > > > > I could just change the default port on asterisk to another one, > but I > > > > would also like it to bind in 5060 in addition to the extra ports > for > > > > "special isps" > > > > > > > > The ISP blocking the port 5060 is Airtel in Bangalore, India, for > the > > > > curiosity of others.. > > > > > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > > > Are you in the USA? > > > > > > > > > > If so call the FCC, they do not like port 5060 blocking (or any > > > other > > > > > VoIP port blocking) > > > > > > > > > > See here: http://www.google.com/search?hl=en&q=fcc+fine+voip > > > > > > > > > > Not the technical answer you are looking for but the RIGHT > answer. > > > > > > > > > > > -Original Message- > > > > > > From: [EMAIL PROTECTED] > > > [mailto:asterisk-users- > > > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > > > To: asterisk-users@lists.digium.com > > > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > > > > > Hello all > > > > > > > > > > > > I'm trying to get my asterisk config to listen to multiple > ports. > > > This > > > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > > > supported? > > > > > > > > > > > > Thanks! > > > > > > ___ > > > > > > Asterisk-Users mailing list > > > > > > Asterisk-Users@lists.digium.com > > > > > > 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
I could, but I don't want to do that. I just want to be able to have asterisk listen on multiple SIP ports. Is that possible? No need to keep trying to go around the issue.. On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > Or you could run another * box as a "proxy" (proxy is not the right > term, but you get the idea) on different port and trunk to your main box > with iax. > > > -Original Message- > > From: Prepaid [mailto:[EMAIL PROTECTED] > > Sent: Monday, June 13, 2005 10:54 PM > > To: Damon Estep; asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > The end point that is having the issue is not in the US. > > > > I could just change the default port on asterisk to another one, but I > > would also like it to bind in 5060 in addition to the extra ports for > > "special isps" > > > > The ISP blocking the port 5060 is Airtel in Bangalore, India, for the > > curiosity of others.. > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > Are you in the USA? > > > > > > If so call the FCC, they do not like port 5060 blocking (or any > other > > > VoIP port blocking) > > > > > > See here: http://www.google.com/search?hl=en&q=fcc+fine+voip > > > > > > Not the technical answer you are looking for but the RIGHT answer. > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > To: asterisk-users@lists.digium.com > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > Hello all > > > > > > > > I'm trying to get my asterisk config to listen to multiple ports. > This > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > supported? > > > > > > > > Thanks! > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
The RTP ports are not blocked, I have already binded on the other port and it works fine, I would just like to have Asterisk bind to multiple ports. If they start blocking the 10,000 to 20,000 ports for RTP it will never end and they might as well not allow connections on any out going ports besides 80. On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > You will not win, once you get past the 5060 issue you might find RTP > blocked also. > > > > -Original Message- > > From: Prepaid [mailto:[EMAIL PROTECTED] > > Sent: Monday, June 13, 2005 10:54 PM > > To: Damon Estep; asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] SIP Listen to multiple ports > > > > The end point that is having the issue is not in the US. > > > > I could just change the default port on asterisk to another one, but I > > would also like it to bind in 5060 in addition to the extra ports for > > "special isps" > > > > The ISP blocking the port 5060 is Airtel in Bangalore, India, for the > > curiosity of others.. > > > > On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > > > Are you in the USA? > > > > > > If so call the FCC, they do not like port 5060 blocking (or any > other > > > VoIP port blocking) > > > > > > See here: http://www.google.com/search?hl=en&q=fcc+fine+voip > > > > > > Not the technical answer you are looking for but the RIGHT answer. > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Prepaid > > > > Sent: Monday, June 13, 2005 10:46 PM > > > > To: asterisk-users@lists.digium.com > > > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > > > > > Hello all > > > > > > > > I'm trying to get my asterisk config to listen to multiple ports. > This > > > > is since some clients have port 5060 blocked by their ISP. > > > > > > > > Does anyone know how to do this in sip.conf or if it is even > > > supported? > > > > > > > > Thanks! > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
The end point that is having the issue is not in the US. I could just change the default port on asterisk to another one, but I would also like it to bind in 5060 in addition to the extra ports for "special isps" The ISP blocking the port 5060 is Airtel in Bangalore, India, for the curiosity of others.. On 6/14/05, Damon Estep <[EMAIL PROTECTED]> wrote: > Are you in the USA? > > If so call the FCC, they do not like port 5060 blocking (or any other > VoIP port blocking) > > See here: http://www.google.com/search?hl=en&q=fcc+fine+voip > > Not the technical answer you are looking for but the RIGHT answer. > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Prepaid > > Sent: Monday, June 13, 2005 10:46 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] SIP Listen to multiple ports > > > > Hello all > > > > I'm trying to get my asterisk config to listen to multiple ports. This > > is since some clients have port 5060 blocked by their ISP. > > > > Does anyone know how to do this in sip.conf or if it is even > supported? > > > > Thanks! > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > 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 Asterisk-Users@lists.digium.com 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] SIP Listen to multiple ports
Are you in the USA? If so call the FCC, they do not like port 5060 blocking (or any other VoIP port blocking) See here: http://www.google.com/search?hl=en&q=fcc+fine+voip Not the technical answer you are looking for but the RIGHT answer. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Prepaid > Sent: Monday, June 13, 2005 10:46 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] SIP Listen to multiple ports > > Hello all > > I'm trying to get my asterisk config to listen to multiple ports. This > is since some clients have port 5060 blocked by their ISP. > > Does anyone know how to do this in sip.conf or if it is even supported? > > Thanks! > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Listen to multiple ports
Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] octtel SP 4220 gateway and Asterisk
Did you ever get your SP4220 going? On 5/12/05, scott <[EMAIL PROTECTED]> wrote: > > > > Hi Peoples > > > > I would be interested to hear from anyone who has managed to get the Octtel > SP4220 and asterisk talking together. > > > > I am using the Octtel as a gateway for a PSTN line. It passes the call on to > Asterisk and then Asterisk moves the call to a particular extension. Whilst > I can get it to do this, there is no sound. It appears that Asterisk is > trying to create a "native bridge" between the gateway and the sip > extension. But this fails, or connects but again with no sound. > > > > I would appreciate any help at all. You are welcome to email msn/yahoo me > > > > Regards > > > > Scott k > > > > [EMAIL PROTECTED] > > > > [EMAIL PROTECTED] (MSN) > > > > [EMAIL PROTECTED] > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] ztcfg server crash
I tried to get * stable on a 2.6xxx kernel for about 2 weeks. Then tried it out on a FC1 2.4.xxx kernel and found none of the issues. I am sure others have had success with > 2.4.xxx, but I gave up;) BTW - I was using a TE110P and then a TE405P card for the zaptel install. Both were setup as T1s not E1s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztcfg server crash I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: > > What OS/distro are you running? > > I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to > FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Rod > Bacon > Sent: Monday, June 13, 2005 7:31 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] ztcfg server crash > > I was wondering if anyone had experienced the following with asterisk > stable. > > After a period of time (can vary), If I stop asterisk and try to run > ztcfg -v to reinitialise my quad e1 card, the server will lock up. > Sometimes it's a complete lockup, where it won't even return pings, > other times it seems to be "partially screwed". > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 6/11/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
This is 99% a "users" issue. Provocative, important reading. (it might also be "dev" but not "biz") Keep it here. This is a very interesting conversation, but it seems like the BIZ forummight be more appropriate. Jason Sjobeck ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 connecting remote villages in western Uganda
Mark, You are a gentleman & a scholar. This is tremendous. I wish sometimes I was a hippie so I could run barefoot with all of humankind (ie: do these kinds of projects), but, alas, I was raised during the Gordon Gecko era. There are just not enough of "you" out there. I wish there more. I want to contribute. For others out there looking for a meaningful way to contribute, PayPal over a few bucks and see if you can convince your employer to PayPal his organization $100 to do more of these. Superb. Jason I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa. Mark Mark Summer co-founder, Inveneo web: http://www.inveneo.org phone: +1-415-901-1969 x 1200 FWD: 603303 cell: +1-415-867-9751 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help connecting two * pcs with *@home
On 6/13/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> What I want to be able to do is dial ext 2009 from server b (2009 doesnt exist
> on this server) and have it ring 2009 on server a. If that is difficult to do,
> I can always revert to dialing a prefix, eg. 22009 to call server a.
If you're using IAX, you may find the following of some use:
Server A:
In iax.conf:
[general]
register => andy:[EMAIL PROTECTED]
[andy]
type=friend
context=otherservervoip
auth=rsa
inkeys=otherserver
In extensions.conf:
[otherservervoip]
exten = _66XXX,1,Dial(IAX2/andy:[EMAIL PROTECTED]/${EXTEN:2})
Server B:
[andy]
type=friend
secret=mypassword
username=andy
host=dynamic
context=default
outkey= otherservervoip
I gleaned most of this from the wiki (voip-info.org). Highly
recommended and valuable resource. Be prepared to play around with
this stuff a bit first before it works. I think you'll find the
documentation extremely helpful.
-Andy
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Re: [Asterisk-Users] Group/Broadcast Voicemail
I did change char tmp[4096], *ext; to 4096 but there's also the same
line under vm_execmain but I really don't know anything about
programming. I only saw the same line.
Robert Goodyear wrote:
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Group/Broadcast Voicemail
No, I can't get characters below 256.
Robert Goodyear wrote:
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Correct me if I am wrong. I can remember installing a T1's with a HDSL unit at the last CO, in which the T1 was delivered to the customer's prem in two wires. I think they called this fast half-duplex. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Coulson Sent: Monday, June 13, 2005 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Leon Sun wrote: > Not really true about T1 description. When you apply for T1, you need tell > vendor if it's channelized or non-ch. If you are going to use it for 1.5M > network, you need use unchannelized T1. T1 is T1. How you use the DS0s delivered across it is up to you. You can mux them out to POTS lines, use them all for data or mix it up and run voice and data over the same T1. Telco vendors don't care what you do with it, unless it's terminating for data/voice in their equipment. Even when you use all 24 channels for data, they still function as 24 distinct DS0 channels as far as timing is concerned. Unlike OC-nc circuits (Where you save some overhead for the sake of being unable to channelize the STS channels) , there is no overhead variation when channelizing a DS-1 versus using a full DS-1 for data. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] problem with pf and asterisk
I took the info from here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules and ended up with the following in my pf.conf: rdr on $ext_if proto tcp from any to ($ext_if) port 5060 -> $dmz_ip port 5060 rdr on $ext_if proto udp from any to ($ext_if) port 5060 -> $dmz_ip port 5060 rdr on $ext_if proto udp from any to ($ext_if) port 4569 -> $dmz_ip port 4569 rdr on $ext_if proto udp from any to ($ext_if) port 5036 -> $dmz_ip port 5036 rdr on $ext_if proto udp from any to ($ext_if) port :20001 -> $dmz_ip port :20001 rdr on $ext_if proto udp from any to ($ext_if) port 2727 -> $dmz_ip port 2727 I also have the following lines in there: pass out on $ext_if all keep state # pass incoming dmz traffic pass in on $ext_if proto tcp from any to $dmz_ip keep state pass in on $ext_if proto udp from any to $dmz_ip keep state HTH Hatton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Authentication
Title: Message Race, Are you saying that the default is autocreatepeers=yes? I was under the impression that the default is no and yes must be explicitly defined. Same holds true for insecure=, default no, optional yes or very. Please tell me I am not mistaken so I do not feel compelled to review a years worth of telecom bills line by line J Damon Greetings, You have stumbled on to one of the most troublesome flag for newbies; autocreatepeer. http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer in your sip.conf file add a line in the [general] section “autocreatepeer=no” Now people can only use your Asterisk SIP connection if you create a peer entry for them in your sip.conf file. Your sip.conf file should be located in “/etc/asterisk” directory. “cd /etc/asterisk” “vi sip.conf” ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ztcfg server crash
I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ztcfg server crash
What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 6/11/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
On Mon, 13 Jun 2005, Damon Estep wrote: You guys have me second guessing my training and experience in this area, so; 1. If I am wrong I apologize to the group. 2. I have been trying for a few minutes to find confirmation either way. From what I know about the modulation techniques used by DSL (DMT, CAP, QAM) it is impossible for the transceiver in the device to transmit and receive at the same time (unless there is discreet "channels" for each path and a very good transceiver). There are three frequency ranges on ADSL, the voice portion (lowest), the upstream (middle), and the downstream (highest). They are split into different frequencies so filters can pick out the up and down. It's capable of full duplex, but as the downstream speed increases, the big differential causes the issue below. Can anyone provide a more logical explanation of why the outbound latency on every DSL modem tested increases with inbound traffic? Even at rates well below the maximum data rate, Not the case on a T1. My explanation is that the additional latency is due to packet scheduling and queuing mechanisms required by the technology. That is usually due to the TCP acks getting delayed in transit, making the other end wait to send the next packets. On a symmetrical circuit you don't see it as much, but the asymmetry on ADSL makes it more apparent. dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Are you sure? Everything I have seen says SDSL = Full Duplex. That being achieved by dropping the pair that provided voice and using it for signalling. "Where ADSL utilizes unoccupied frequencies and averts conflict with analog voice frequencies, SDSL takes over the whole line. SDSL eliminates analog voice capabilities in favor of full-duplex data transmission. No splitter, no analog voice-nothing but data. As a decent alternative to T1, SDSL has gotten a fair amount of attention from Competitive Local Exchange Carriers." Excerpt from http://www.isp-select.com/SDSL.htm Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing
[Asterisk-Users] ztcfg server crash
I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Notice below that the only forms of DSL touted by anyone as replacements for full duplex 1.544mbps T1 lines is HDSL. Telcos regularly use HDSL as replacements for traditional DS1 service wher the line distances are VERY short, in most cases the HDSL circuit requires 2 pair, in some very short line distance installs it can be doen with one pair. HDSL interfaces cost 5 times what typical DSL interfaces cost. If you could get full 1.544mbps full duplex emulation off of sdsl why is there no one doing it? Sure would be a lot cheaper to provision. Have you ever gotten 22 simultaneous g.711 calls to run on a 1.5mbps dsl line? (I said 22, not 24, leaving 128kbps for signaling and protocol overhead). My experience and testing sets the real would limits to 20 to 21 on a T1 and 10 to 12 on a 1.5M SDSL circuit. I am not sure that I am that wrong in my original reply, unless there is someone here with more definitive technical references. Never one to take on an entire mailing list, I just want to be sure the next time I spout off about DSL I am certain I know what I am talking about. Live and learn, right? Damon Q: What are the various types of xDSL? A: There are several forms of xDSL, each designed around specific goals and needs of the marketplace. Some forms of xDSL are proprietary, some are simply theoretical models and some are widely used standards. They may best be categorized within the modulation methods used to encode data. Below is a brief summary of some of the known types of xDSL technologies. ADSL Asymmetric Digital Subscriber Line (ADSL) is the most popular form of xDSL technology. The key to ADSL is that the upstream and downstream bandwidth is asymmetric, or uneven. In practice, the bandwidth from the provider to the user (downstream) will be the higher speed path. This is in part due to the limitation of the telephone cabling system and the desire to accommodate the typical Internet usage pattern where the majority of data is being sent to the user (programs, graphics, sounds and video) with minimal upload capacity required (keystrokes and mouse clicks). Downstream speeds typically range from 768 Kb/s to 9 Mb/s Upstream speeds typically range from 64Kb/s to 1.5Mb/s. ADSL Lite (see G.lite) CDSL Consumer Digital Subscriber Line (CDSL) is a proprietary technology trademarked by Rockwell International. CiDSL Globespan's proprietary, splitterless Consumer-installable Digital Subscriber Line (CiDSL). EtherLoop EtherLoop is currently a proprietary technology from Nortel, short for Ethernet Local Loop. EtherLoop uses the advanced signal modulation techniques of DSL and combines them with the half-duplex "burst" packet nature of Ethernet. EtherLoop modems will only generate hi-frequency signals when there is something to send. The rest of the time, they will use only a low-frequency (ISDN-speed) management signal. EtherLoop can measure the ambient noise between packets. This will allow the ability to avoid interference on a packet-by-packet basis by shifting frequencies as necessary. Since EtherLoop will be half-duplex; it is capable of generating the same bandwidth rate in either the upstream or downstream direction, but not simultaneously. Nortel is initially planning for speeds ranging between 1.5Mb/s and 10Mb/s depending on line quality and distance limitations. G.lite A lower data rate version of Asymmetric Digital Subscriber Line (ADSL) was been proposed as an extension to ANSI standard T1.413 by the UAWG (Universal ADSL Working Group) led by Microsoft, Intel, and Compaq. This is known as G.992.2 in the ITU standards committee. It uses the same modulation scheme as ADSL (DMT), but eliminates the POTS splitter at the customer premises. As a result, the ADSL signal is carried over all of the house wiring which results in lower available bandwidth due to greater noise impairments. Often a misnomer, this technology is not splitterless per se. Instead of requiring a splitter at customer premises, the splitting of the signal is done at the local CO. G.shdsl G.shdsl is an ITU standard which offers a rich set of features (e.g. rate adaptive) and offers greater reach than many current standards. G.shdsl also allows for the negotiation of a number of framing protocols including ATM, T1, E1, ISDN and IP. G.shdsl is touted as being able to replace T1, E1, HDSL, SDSL HDSL2, ISDN and IDSL technologies. HDSL High Bit-rate Digital Subscriber Line (HDSL) is generally used as a substitute for T1/E1. HDSL is becoming popular as a way to provide full-duplex symmetric data communication at rates up to 1.544 Mb/s (2.048 Mb/s in Europe) over moderate distances via conventional telephone twisted-pair wires. Traditional T1 (E1 in Europe) requires repeaters every 6000 ft. to boost the signal strength. HDSL has a longer range than T1/E1 without the use of repeaters to allow transmission over distances up to 12,000 feet. It uses pulse amplitude modulation (PAM) on a 4-wire loop. HDSL2 High Bit-rate Digital
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
OK, You guys have me second guessing my training and experience in this area, so; 1. If I am wrong I apologize to the group. 2. I have been trying for a few minutes to find confirmation either way. >From what I know about the modulation techniques used by DSL (DMT, CAP, QAM) it is impossible for the transceiver in the device to transmit and receive at the same time (unless there is discreet "channels" for each path and a very good transceiver). Does anyone have any definitive technical resources confirming that any form of xDSL technology can transmit and receive at precisely the same time (not interleaved). Can anyone provide a more logical explanation of why the outbound latency on every DSL modem tested increases with inbound traffic? Even at rates well below the maximum data rate, Not the case on a T1. My explanation is that the additional latency is due to packet scheduling and queuing mechanisms required by the technology. Maybe I will learn something this evening. Damon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Are you sure? Everything I have seen says SDSL = Full Duplex. That being achieved by dropping the pair that provided voice and using it for signalling. "Where ADSL utilizes unoccupied frequencies and averts conflict with analog voice frequencies, SDSL takes over the whole line. SDSL eliminates analog voice capabilities in favor of full-duplex data transmission. No splitter, no analog voice-nothing but data. As a decent alternative to T1, SDSL has gotten a fair amount of attention from Competitive Local Exchange Carriers." Excerpt from http://www.isp-select.com/SDSL.htm Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: > Barton Fisher wrote: >> I'm looking to expand my bandwidth for my Asterisk PBX. Why should I >> choose a T1 over DSL for my asterisk server? I found someone >> offering T1's for $290 a month + Loops or 3 Meg for $561 a month + >> Loops. Is this a good deal? >> Thanks >> Bart >> - >> - >> -- >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > Where are you located? What CLEC gives you a T-1 for $290? > FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/ma
[Asterisk-Users] Need help connecting two * pcs with *@home
I have read through the info available on the list and searched for things with google. I have found quite a bit of info on the subject, mainly to do with 'simply add this to iax.conf and this to extentions.conf and it will work'. So I tried all of those things and I can't for the life of me get this thing to work properly. What I want to be able to do is dial ext 2009 from server b (2009 doesnt exist on this server) and have it ring 2009 on server a. If that is difficult to do, I can always revert to dialing a prefix, eg. 22009 to call server a. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reading about G729
Greetings! Please, somebody give me a link to read a good article about G729 on the web or pdf. (buying it from Digium and installing, advantages and disadvantages etc...etc..) Thank you Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Digium Website Update: Asterisk Business Edition
Hi List. I think we as users have a choice to look at this a bit differently. You as a user, consultant or reseller of asterisk services, simply have more options here. It also makes it much easier to implement asterisk in the enterprise to be able to provide it with guaranteed support. I for my part, now can choose to install Asterisk and support it myself, with access to the community, or I can install Asterisk and buy support directly from Digium. Guaranteeing quality and support is quite hard while providing source code, Digium could easily spend countless hours on a bug implemented by their customer (and the customer would furiously deny that he had made any changes). It wouldn't make any sense for digium to implement features only in the proprietary license spectrum, when they can do it and get it tested opensource for free, unless there is a really big enterprise ready to pay to iron out the bugs and test it. No complaints from me here as long as their sticking to the same source (as Kevin says clearly below). Just as well I think there are mailing-lists available on the internet to discuss various licensing options. Maron Kristofersson Kevin P. Fleming wrote: > Aidan Van Dyk wrote: > >> In either case, since they are committed to the open source model, are >> they willing to tell us what features/bug fixes in CVS are considered an >> increased liability and risk, or what important features and bug fixes >> they've applied on top of CVS? This could help those of us trying to >> build stable, robust, asterisk-based solutions promote asterisk as a >> stable, robust, low-risk platform. > > > There is absolutely _no_ code in ABE that is not in CVS HEAD, except for > the code related to license control required for the commercial product > and an installer (since it's a binary distribution). We have not, and > will not, put any bug fixes or new features into ABE that do not go into > CVS HEAD as well (and in fact I don't think there's any situation where > a feature would go into ABE first). > > Simply stated, ABE is a supported, documented, tested and commercially > packaged/licensed version of Asterisk, designed for companies who want > such a package. It is missing quite a number of features from CVS HEAD > that were deemed unnecessary and/or difficult to support, and it will > not be updated as often as CVS HEAD itself is (or even as often as the > stable releases are made), from what I understand. It does not come with > source code, nor is the exact source code available to anyone outside of > Digium, which is done purely for support reasons. ABE users can call > Digium support and the support staff will know _exactly_ what they are > running, since it is a packaged release. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Issues...
Hello all I have looked in Google for a fix for this issue but I have had no luck, so I am posting it here. Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting causecode to be single byte but was 35 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting samplerate to be 2 bytes long but was 40 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting samplerate to be 2 bytes long but was 40 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting block desc to be 4 bytes long but was 41 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting block desc to be 4 bytes long but was 41 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Expecting callingtns to be 2 bytes long but was 42 Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:625 iax_error_output: Information element length exceeds message size Jun 10 03:42:17 WARNING[2961]: Chan_iax2.c:6565 socket_read: Undecodable frame received from ‘XX.XX.XX.XX’ Any help on debugging this would be great. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Leon Sun wrote: > Not really true about T1 description. When you apply for T1, you need tell > vendor if it's channelized or non-ch. If you are going to use it for 1.5M > network, you need use unchannelized T1. T1 is T1. How you use the DS0s delivered across it is up to you. You can mux them out to POTS lines, use them all for data or mix it up and run voice and data over the same T1. Telco vendors don't care what you do with it, unless it's terminating for data/voice in their equipment. Even when you use all 24 channels for data, they still function as 24 distinct DS0 channels as far as timing is concerned. Unlike OC-nc circuits (Where you save some overhead for the sake of being unable to channelize the STS channels) , there is no overhead variation when channelizing a DS-1 versus using a full DS-1 for data. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Not really true about T1 description. When you apply for T1, you need tell vendor if it's channelized or non-ch. If you are going to use it for 1.5M network, you need use unchannelized T1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: June 13, 2005 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Hi David, You are correct, I always get those 2 confused. Thanks for the clearing. Nir S David Coulson wrote: >Nir Simionovich wrote: > > >> Now, E1 and T1 lines are based upon a channel based connection, which >>means you get a line >>with X number of data lines and a single control/signalling line. On T1 >>it means that you have 23 >>lines dedicated for Voice/Data (each is 64kbps) and a single signaling >>line (64kbps). >> >> > >A T1 has no seperate signaling line - You're thinking of PRI. T1 gives >you 24 DS0 (64kbit) channels, which you can do whatever you want with. >PRI just shanks off one channel for D channel signaling. > >David > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Hi David, You are correct, I always get those 2 confused. Thanks for the clearing. Nir S David Coulson wrote: Nir Simionovich wrote: Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
BTW - The Speakeasy SDSL connection I originally posted about is
delivered via Covad.
The SLA (some of it at least)
Average Network Delivery and Delay2 - Further proof that Covad has
confidence in the performance of our network.
Delivery - 99.9% successful delivery of all data packets sent from your
location over the Covad network, or you will be eligible for a credit of
up to 10% of your monthly service fee.
Delay - 110 millisecond average for the round trip of a message sent
from your location to a test point on the Covad network, or you will be
eligible for a credit of up to 10% of your monthly service fee.
SLA can be found here...
http://www.covad.com/products/access/telespeed/details.shtml#sla
Being only 4000 feet from the Central Office, this works very well for
me.
I have not been able to figure if QoS is possible yet.
Haven't figured out the examples from the Wiki for QoS via HFB (I think)
and no answer from techs yet.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: Monday, June 13, 2005 6:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Marcelo Pacheco wrote:
>SDSL has symmetrical speeds and full duplex communications.
>Of the widely deployed lan/wan technologies, the only one I know of
>that is half-duplex is 802.11{b,g}.
>
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or asymetrics of the actual physical line.
>The only technical difference between a T1 and SDSL is how it's
>physically delivered to the customer, what usually happens is that a T1
>is not oversold, while an SDSL is oversold anywhere from 8:1 to 3:1.
>
>
That is correct in the genereal idea, however, as xDSL technologies are
switched technologies, unlike cable (DOCSIS) technologies, the fact that
you are overloaded 8:1 or 3:1 will not really matter. As long as your
equipment supports QoS correctly, you shouldn't have a problem.
>ADSL is full duplex as well, if you don't know how to do QOS then it
>will feel like it's half duplex, but it's not. I have 1000/320 ADSL
>that I can use full bandwidth both ways.
>
>
ADSL appears to be half-duplex only due to the fact that most ISP's
"misconfigure" the modems and routers.
As a rule of thumb, the modem/router can be re-configured to utilize
both channels to the fullest, but again, this must rely on the fact that
your ISP's equipment supports QoS at the switch level correctly.
Nir S
>Marcelo Pacheco
>
>Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
>
>
>>You are aware that DSL (even SDSL) is half duplex and a T1 is full
>>duplex, right?
>>
>>1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
>>1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
>>
>>This is due to a separate transmit and receive path on a t1 and a
>>shared path on sdsl.
>>
>>The s in sdsl means symmetrical, not duplex, that is that the
>>signaling rate is the same in either direction, but still half duplex.
>>
>>For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
>>nature of the traffic, unlike most internet that is download-centric.
>>
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
>>Siler
>>Sent: Monday, June 13, 2005 9:43 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
>>
>>Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
>>only costs around $100 per month.
>>
>>W
>>
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of Robert
>>Goodyear
>>Sent: Friday, June 10, 2005 7:02 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
>>
>>On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
>>
>>
>>>Barton Fisher wrote:
>>>
>>>
I'm looking to expand my bandwidth for my Asterisk PBX. Why should
I
choose a T1 over DSL for my asterisk server? I found someone
offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
Loops. Is this a good deal?
Thanks
Bart
-
-
--
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>>>Where are you located? What CLEC gives you a T-1 for $290?
>>>
>>>
>>FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
>>getting a break for having a voice and a data circuit broken out from
>>one fiber drop, but that's what I'm payi
RE: [Asterisk-Users] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance Sprint nevermore... I switched voer to Sprint a few years ago and they literally dropped service form under us. It was during that Sprint ION fiasco. They sold to me, installed, and literally terminated the service 10 days later. The whole time they were working the install for me, the other side of the company was going belly up. Poo poo on sprint... w From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel BatistaSent: Monday, June 13, 2005 5:31 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] MCI vs. XO/Allegiance we have been using XO/Allegiance for over 3 years and have had no problems. I can't compare to MCI but we also had a sprint t1 that we had to get remove due to them being bad in billing and also not very reliable for faxing. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 13, 2005 6:59 PM Subject: [Asterisk-Users] MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Nir Simionovich wrote: > Now, E1 and T1 lines are based upon a channel based connection, which > means you get a line > with X number of data lines and a single control/signalling line. On T1 > it means that you have 23 > lines dedicated for Voice/Data (each is 64kbps) and a single signaling > line (64kbps). A T1 has no seperate signaling line - You're thinking of PRI. T1 gives you 24 DS0 (64kbit) channels, which you can do whatever you want with. PRI just shanks off one channel for D channel signaling. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance we have been using XO/Allegiance for over 3 years and have had no problems. I can't compare to MCI but we also had a sprint t1 that we had to get remove due to them being bad in billing and also not very reliable for faxing. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 13, 2005 6:59 PM Subject: [Asterisk-Users] MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
Marcelo Pacheco wrote:
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or
asymetrics of the actual physical line.
The only technical difference between a T1 and SDSL is how it's physically
delivered to the customer, what usually happens is that a T1 is not oversold,
while an SDSL is oversold anywhere from 8:1 to 3:1.
That is correct in the genereal idea, however, as xDSL technologies are
switched technologies, unlike
cable (DOCSIS) technologies, the fact that you are overloaded 8:1 or 3:1
will not really matter. As long
as your equipment supports QoS correctly, you shouldn't have a problem.
ADSL is full duplex as well, if you don't know how to do QOS then it will feel
like it's half duplex, but it's not. I have 1000/320 ADSL that I can use full
bandwidth both ways.
ADSL appears to be half-duplex only due to the fact that most ISP's
"misconfigure" the modems and routers.
As a rule of thumb, the modem/router can be re-configured to utilize
both channels to the fullest, but again, this
must rely on the fact that your ISP's equipment supports QoS at the
switch level correctly.
Nir S
Marcelo Pacheco
Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a shared
path on sdsl.
The s in sdsl means symmetrical, not duplex, that is that the signaling
rate is the same in either direction, but still half duplex.
For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
nature of the traffic, unlike most internet that is download-centric.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
Barton Fisher wrote:
I'm looking to expand my bandwidth for my Asterisk PBX. Why should I
choose a T1 over DSL for my asterisk server? I found someone
offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
Loops. Is this a good deal?
Thanks
Bart
-
-
--
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Where are you located? What CLEC gives you a T-1 for $290?
FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
getting a break for having a voice and a data circuit broken out from
one fiber drop, but that's what I'm paying here in Orange County. Also,
I had a business cable modem before, which was *allegedly* not shared
for business customers (suspicious) and the throughput was a roller
coaster, as was the latency. The DS-1 cleared all that up.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Damon, I have no idea where you are getting your information from, but what you said makes no sense. DSL based lines, be it ADSL or SDSL, are based upon a connection technology in the ATM family. As a result, the upstream and downstream of the connection can be controlled seperately. If someone offers you a 1.5 SDSL connection, it doesn't actually mean that you have 2x768kbps, it may actually mean that you have 2x1.5Mbps. However, that speed is only towards your internet provider, what you get beyond that point would be bound to your ISP's SLA and contract. Now, E1 and T1 lines are based upon a channel based connection, which means you get a line with X number of data lines and a single control/signalling line. On T1 it means that you have 23 lines dedicated for Voice/Data (each is 64kbps) and a single signaling line (64kbps). Now, lets do a little math (23+1)*64 = 1536kbps = 1.536Mbps, hence the speed for a single T1 circuit. Now, if you have a T1 installed, and you are currently using 512kbps of upload, it means that you are physically using 8 lines out of the 23 data lines for uploading. You can then use the rest to what ever purpose you want, but while those lines are in play, you won't be upload another 512kbps on the same lines. The reason for that is that each of these lines operates on a seperate Time Slot within the physical layer. Once a Time Slot is taken for a specific data flow, it can't be used for another data flow. This actually means that a T1 will give you a shared 1.5Mbps towards your ISP, with speed that vary on the upload and download, according to your usage. While when using a DSL, your quality of service for the connection to the ISP is described by the policy of connection. In many countries (eg: Israel, Turkey, China, UK), DSL lines are actually ADSL lines, where the downstream is around 1.5Mbps while the uplink is around 128kbps (just enough to do a little VoIP). Last time I was in the UK, about 4 weeks ago, I noticed they are now selling 8Mbps ADSL connection to your house, however, the uplink is 512kbps. I would suggest that you get all the information from your providers regarding the type of services rendered on the SDSL line, and make sure that it's the right one for you. Nir S Damon Estep wrote: You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart - - -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/ast
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Are you sure? Everything I have seen says SDSL = Full Duplex. That being achieved by dropping the pair that provided voice and using it for signalling. "Where ADSL utilizes unoccupied frequencies and averts conflict with analog voice frequencies, SDSL takes over the whole line. SDSL eliminates analog voice capabilities in favor of full-duplex data transmission. No splitter, no analog voice-nothing but data. As a decent alternative to T1, SDSL has gotten a fair amount of attention from Competitive Local Exchange Carriers." Excerpt from http://www.isp-select.com/SDSL.htm Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, June 13, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: > Barton Fisher wrote: >> I'm looking to expand my bandwidth for my Asterisk PBX. Why should I >> choose a T1 over DSL for my asterisk server? I found someone >> offering T1's for $290 a month + Loops or 3 Meg for $561 a month + >> Loops. Is this a good deal? >> Thanks >> Bart >> - >> - >> -- >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > Where are you located? What CLEC gives you a T-1 for $290? > FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
The only technical difference between a T1 and SDSL is how it's physically
delivered to the customer, what usually happens is that a T1 is not oversold,
while an SDSL is oversold anywhere from 8:1 to 3:1.
ADSL is full duplex as well, if you don't know how to do QOS then it will feel
like it's half duplex, but it's not. I have 1000/320 ADSL that I can use full
bandwidth both ways.
Marcelo Pacheco
Em Seg 13 Jun 2005 20:54, Damon Estep escreveu:
> You are aware that DSL (even SDSL) is half duplex and a T1 is full
> duplex, right?
>
> 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
> 1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
>
> This is due to a separate transmit and receive path on a t1 and a shared
> path on sdsl.
>
> The s in sdsl means symmetrical, not duplex, that is that the signaling
> rate is the same in either direction, but still half duplex.
>
> For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex
> nature of the traffic, unlike most internet that is download-centric.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wiley
> Siler
> Sent: Monday, June 13, 2005 9:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
>
> Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
> only costs around $100 per month.
>
> W
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Robert
> Goodyear
> Sent: Friday, June 10, 2005 7:02 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
>
> On Jun 10, 2005, at 6:38 PM, Michael Welter wrote:
> > Barton Fisher wrote:
> >> I'm looking to expand my bandwidth for my Asterisk PBX. Why should I
> >>
> >> choose a T1 over DSL for my asterisk server? I found someone
> >> offering T1's for $290 a month + Loops or 3 Meg for $561 a month +
> >> Loops. Is this a good deal?
> >> Thanks
> >> Bart
> >> -
> >> -
> >> --
> >> ___
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > Where are you located? What CLEC gives you a T-1 for $290?
>
> FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm
> getting a break for having a voice and a data circuit broken out from
> one fiber drop, but that's what I'm paying here in Orange County. Also,
> I had a business cable modem before, which was *allegedly* not shared
> for business customers (suspicious) and the throughput was a roller
> coaster, as was the latency. The DS-1 cleared all that up.
>
> /rg
>
> Robert Goodyear
> Brand Up LLC
> http://www.brand-up.com
>
> ___
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[Asterisk-Users] problem with pf and asterisk
current setup SIP phone 192.168.1.30 --> linksys wrt54g sveasoft -- INTERNET -- (xl0) Firewall (xl2:172.16.0.50)--> (em1:172.16.0.101) Asterisk problem is RTP stream not oging trouhg from * to sip and vice versa. #1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as return address or #2 asterisk trying to get back to me as 192.168 on public internet.. got canreinvite=yes and no. nat=yes qualify=1000 externaladdr=IP of (em1) localnet=172.16.0.0/12 i would need help form someone who did a sismilar setup.. i do run carp and pfsync also on the FW. mirrored to FW2 down ATM... anyhelp appreciated.. banging head on the wall for 2 weeks now.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Should I choose DSL @ 1.5 or a full T1?
You are aware that DSL (even SDSL) is half duplex and a T1 is full duplex, right? 1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out 1.5 in. a T1 will do 1.5 in and 1.5 out sustained. This is due to a separate transmit and receive path on a t1 and a shared path on sdsl. The s in sdsl means symmetrical, not duplex, that is that the signaling rate is the same in either direction, but still half duplex. For VoIP a t1 is worth double what a 1.5 sdsl is because of the duplex nature of the traffic, unlike most internet that is download-centric. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, June 13, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and only costs around $100 per month. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Friday, June 10, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: > Barton Fisher wrote: >> I'm looking to expand my bandwidth for my Asterisk PBX. Why should I >> choose a T1 over DSL for my asterisk server? I found someone >> offering T1's for $290 a month + Loops or 3 Meg for $561 a month + >> Loops. Is this a good deal? >> Thanks >> Bart >> - >> - >> -- >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > Where are you located? What CLEC gives you a T-1 for $290? > FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
Let me throw another complaint against XO on the table. They actually shut off the wrong T1 and they transferred all of the DIDs to the T1 they shut off! how screwed up is that? We are now about 2 years later and their billing department still calls us every month for nonpayment of the T1 that they turned off. We also have 6 T1s through MCI, all long distance. They have been much better to deal with. MATT--- -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Monday, June 13, 2005 7:34 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MCI vs. XO/Allegiance Well, the fact that two negatives for XO and a positive for MCI all came at once says a lot to me. Interestingly enough their SLA reads... * 24/7/365 Network Monitoring and Service. If for some reason your network is having problems, the chances are XO will know about it before you do and respond before any problems become critical. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue, Jr. Sent: Monday, June 13, 2005 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: > Hello All, > > Anyone out there using ISDN PRI from either MCI or XO/Allegiance? > Gotta make the choice today and the difference per month is only about > $25 in favor of MCI. > > Billing is pretty much the same between the two so I have pretty much no > point of reference on which to choose. > Any thoughts from anyone experienced with these two compnies would be > greatly appreciated! > > Thanks, > Wiley > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
Well, the fact that two negatives for XO and a positive for MCI all came at once says a lot to me. Interestingly enough their SLA reads... * 24/7/365 Network Monitoring and Service. If for some reason your network is having problems, the chances are XO will know about it before you do and respond before any problems become critical. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Traue, Jr. Sent: Monday, June 13, 2005 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: > Hello All, > > Anyone out there using ISDN PRI from either MCI or XO/Allegiance? > Gotta make the choice today and the difference per month is only about > $25 in favor of MCI. > > Billing is pretty much the same between the two so I have pretty much no > point of reference on which to choose. > Any thoughts from anyone experienced with these two compnies would be > greatly appreciated! > > Thanks, > Wiley > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] snom 190: dial tone without registration?
You should use DHCP to enforce IP address to MAC binding when the phones boot. And then let the phones register and use host access (deny/permit) permissions in peer section to restrict by IP address/mask. alan wrote: Gavin Hamill <[EMAIL PROTECTED]> wrote: On Monday 13 June 2005 16:42, alan wrote: I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. enforce SIP channel IP restrictions in Asterisk without "host=", or get the snom 190 to stop complaining when it's not registered How about tackling this with iptables and matching specific IP addresses on specific MAC addresses? This solves part, but not all, of the problem. This ensures that only authorized devices can connect to asterisk, and that their IP addresses are also correct. But it doesn't force each device to use only its assigned sip channel. (That is: with dynamic IP registration, a valid IP/MAC could be configured with another device's SIP registration information, and steal calls which should be going to the other device.) I suppose iptables in combination with sip secrets should be enough. But realistically, I can already do what I want the way I want to do it, with the SPA-841. I mostly need to decide: if this feature is lacking, is it enough for me to prefer the Sipura over the snom? Thanks again, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
I prefer MCI since we use their pri and internet. MCI's support is very pro. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Coulson Sent: June 13, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MCI vs. XO/Allegiance Wiley Siler wrote: > Anyone out there using ISDN PRI from either MCI or XO/Allegiance? We have a DS-3 full of PRI from X/O. They work great, mostly, but their tech support sucks. They screw up number ports all the time and about every week there is some local number I can't dial to via XO which once I open a ticket mysteriously gets fixed without a good explanation. Eventually everything works, but you have to beat on them continously to get things done. Better than dealing with SBC though. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
I'm using an XO pri, and as long as you never change anything on the pri XO's not bad. Our experience is that if you chance anything on the PRI configuration they'll screw it up somehow (YMMV). One thing we have learned is that XO doesn't monitor our voice circuits, so if one of our PRI's goes down, we have to notify them almost immediately or they decommission it so the alert goes away. Paul Wiley Siler wrote: Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] MCI vs. XO/Allegiance
Wiley Siler wrote: > Anyone out there using ISDN PRI from either MCI or XO/Allegiance? We have a DS-3 full of PRI from X/O. They work great, mostly, but their tech support sucks. They screw up number ports all the time and about every week there is some local number I can't dial to via XO which once I open a ticket mysteriously gets fixed without a good explanation. Eventually everything works, but you have to beat on them continously to get things done. Better than dealing with SBC though. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Evening in Melbourne (again!)nextThursday
Make it there if you can! PaulH On Fri, 2005-06-10 at 13:16 +1000, jurgen wrote: > Hi all, > > If you're in Melbourne Australia and interested in Asterisk, you're > invited to join us for the second in an irregularly scheduled casual > evening to talk about Asterisk, VOIP, networks, and just generally get > geeky about IP phone stuff. About a dozen of us got together a couple > of months ago, and had a good time chatting about all things Asterisk. > Beverages were also consumed. > > Anyone with an interest is welcome; from Asterisk Gods to newbies who > have recently downloaded it, from people administering several hundred > seats to people playing with it at home and annoying their families. > > When: Next Thursday evening, the 16th, at 7pm. > Where: Niagara Hotel, 383 Lonsdale Street (between Queen and > Elizabeth) in the city. > > The Niagara's a relaxed, comfortable place, people seemed to like it > last time. Also, like last time, I'll get an old phone and put it on > the table, so those of us who haven't met will be able to recognise > each other. > > Any questions, you can reach me on 0415 276 127, or email > [EMAIL PROTECTED] > > Hope to see you there! > > ...jurgen > > -- > [EMAIL PROTECTED] is jurgen's gmail address. > Visit http://jurgen.ca/ for more yummy goodness. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MCI vs. XO/Allegiance
Title: MCI vs. XO/Allegiance Hello All, Anyone out there using ISDN PRI from either MCI or XO/Allegiance? Gotta make the choice today and the difference per month is only about $25 in favor of MCI. Billing is pretty much the same between the two so I have pretty much no point of reference on which to choose. Any thoughts from anyone experienced with these two compnies would be greatly appreciated! Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voicemail and MS Exchange Synchronization
Yeah, if you get the Microsoft Partners Newsletter emails they reported the 75 GB expansion today. "Increased Storage Limit in Exchange Server Standard Edition Get more out of mission-critical email. In the fall of 2005 the storage limit for Exchange Server 2003 Standard Edition will increase to 75 gigabytes." It took me a while to find it through the links the give you. But here it is http://www.microsoft.com/exchange/downloads/2003/sp2/overview.mspx Then scroll to the bottom of the page... "Mailbox Advancements Drive down operational costs and the complexity of your messaging environments with advances such as: . Increase in mailbox storage size limits to 75 gigabyte (GB) for Exchange Server 2003 Standard Edition in response to customer feedback and evolving mailbox storage needs. . New offline address book format offers significantly improved performance. . Cache mode enforcement with added flexibility. You now can force clients into cached mode to help improve performance and increase the number of active users per server. This is especially beneficial to organizations seeking to further site and server consolidation." -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Brodbeck Sent: Monday, June 13, 2005 4:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization > -Original Message- > From: C F [mailto:[EMAIL PROTECTED] > Sent: Saturday, June 11, 2005 11:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Voicemail and MS Exchange > Synchronization > > > On 6/10/05, Dean Collins <[EMAIL PROTECTED]> wrote: > > Actually I think that has changed to 75gb now (or about to change). > > > Really? any links to support that? Since when is Micro$oft so easy on > giving up on licensing fees? I'm curious, too. If this is true it might save us a lot of pain, upgrade wise. We've been looking at moving away from Exchange entirely because of that damn 16-gig limit, and Exchange Enterprise Edition is just too expensive. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Group/Broadcast Voicemail
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change "in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096]" in an earlier replay so I did.
"static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;"
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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RE: [Asterisk-Users] Re: Digium Website Update: Asterisk BusinessEdition
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > trixter http://www.0xdecafbad.com > Sent: Tuesday, 14 June 2005 3:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Digium Website Update: > Asterisk BusinessEdition > > On Mon, 2005-06-13 at 18:20 +0200, Esben Stien wrote: > > "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> writes: > > > > > Protecting freedoms by putting limits on (thus restricting > > > freedoms). Interesting concept. > > > > I need to repeat here. The gpl's purpose is to protect the freedoms > > that comes with free software. So, you have only the freedoms that > > comes with free software as defined by the FSF. You are not > allowed to > > do what you like. You are constrained to the freedoms that > follows the > > software and I think that is a very interesting concept. > > > > Its not a freedom if its a limit. That is my point. The GPL > doesnt give freedoms it takes them away by putting limits on > other peoples code, not the original authors. Now if the > author is ok with infringing on the rights of others then the > GPL is a good choice, however if the original author is > truely fore freedom in the code process, not the double speak > freedom that FSF talks about (where freedom means taking away > abilities) then they should not follow like sheep and repeat > what the FSF says (which on its face is an outright lie since > its not freedom that it grants). > Ahem, The GPL is about the freedown of the code, not the freedom of the individual > > > > copyright and license to use are different. > > > > I never claimed otherwise. > > If you were the person that was quoting the FSF as fact then you did. > Too bad it got cut out, but you can always go back to the > original post that was claiming freedom means putting limits > on people other than yourself. > > > > > > > You can technically put software out there with no copyright but > > > under the gpl license > > > > Then there would be no one to enforce the license, which > would be bad. > > > > Why do you cut out what I said when I addressed that point? > I am begining to think that you are doing it intentionally now. > > > > > it only restricts *their* code (ie modifications). > > > > Yes, but we also want all modifications to be free > > > > 'we' or you specifically? We is quite a loaded word. The > FSF makes a false claim that it *protects* freedoms, when all > it does is limit the freedoms of others to write code. > Specifically if I take a program and modify it, the original > is still under whatever license I got it in, but > *my* code, the modifications are MINE not the original > authors. The original author has NO right to claim that it > is their work, nor do they have copyright on *my* code. But > by releasing it under a GPL they can force me to use a > license that I may not agree with. This is the reason that I > dont contribute to GPL products, I dont like the idea of > someone else dictating to me how I will distribute *my* code. > > The default GPL makes it a lciense violation to run GPL code > on a commercial (or even BSD) system. Extra stuff has to be > put into the GPL license to say 'its ok if you link this > against non GPL libraries and such'. That is not the > default, so technically unless someone did that putting a > stock GPL license has other limitations on its mere use. At > least historically libc on aix, hpux, sunos (4/5), irix were > all not GPL libc (I dont know with solaris now they added a > bunch of gpl stuff at one point). If any of the GPL licensed > software did not take an overt action to say its ok to run it > on those operating systems then its a license violation. > > That level of selective enforcement also calls into question > the legal standing of the license (if certain sections are > not enforced the whole agreement can be voided on first court > challenge). > > http://www.gnu.org/licenses/gpl-faq.html#TOCLinkingOverControl ledInterface > for linking proprietary code to libraries - overt actions > required to make it work right > > http://www.gnu.org/licenses/gpl-faq.html#TOCGPLCommercially > for reading up on how the license affects others who write code later > > http://www.gnu.org/licenses/gpl-faq.html#TOCGPLIncompatibleAlone > for reading on how you cant really link against libc on a > commercial operating system (or anything with a license that > is not compatible with the GPL, which BSD isnt becuase it > allows someone to take it, write *their own* code in addition > to it and not give *their own* code out. > Thus by default you cant run GPL software on a BSD licensed > system, nor any commercial system *unless* the developer took > an overt action to say this is ok (default GPL it is not ok). > > http://www.gnu.org/licenses/gpl-faq.html#TOCDistributeWithSour ceOn
[Asterisk-Users] VOIP-INFO.ORG website bug
When I try to fing group/call pickup command in www.voip-info.org and made a search like *8 I got an error message. Regards, -- Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Components and suggestions for an asterisk server with 9 to 17 POTS.
A small simple non Dell system and T1/E1 card. If you can even think about 17 lines then start with a PRI. Most PRIs can be ordered as small as 4 lines voice and 768 data that leaves ~8 voice channels open. I can give you system specs off list as the change often. On 6/13/05, Ken Dresdell <[EMAIL PROTECTED]> wrote: > > > > Hello, > > > > What would be the simplest and the cheapest solution to get an Asterisk > server working with 9 to 17 POTS? > > > > Because for 1-8 POTS we are using 1 or 2 Digium TDM cards and past 17 POTS > in our area it is economic to use a PRI. > > > > We are looking for a hardware solution on our side instead of using "did > provider" > > > > Thanks in advance > > > > Ken > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Re: Digium Website Update: Asterisk Busi ness Edition
> -Original Message- > From: Esben Stien [mailto:[EMAIL PROTECTED] > The other problem is the issue that free software developers are > mostly (in my experience) not happy with the fact that their code > would be used in proprietary software. It conflicts with the whole > religion of free software. Well, yeah, that's the whole problem, isn't it? You can't follow the "religion of free software" and still run a company that pays the bills. You have to compromise somewhere. Either you go out of business, or you tick off some of the open source purists. Interesting perspective on this from Forbes: http://www.forbes.com/technology/2005/05/26/cz_dl_0526linux.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition
> -Original Message- > From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] > On Monday 13 June 2005 12:38, The VoIP Connection wrote: > > This is a very interesting converation, but it seems like > the BIZ forum > > might be more appropriate... > > How on earth is this a business-related discussion? -dev > would have been my > guess. :-) Maybe we need an anti-biz list for this kind of thing. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voicemail and MS Exchange Synchronization
> -Original Message- > From: C F [mailto:[EMAIL PROTECTED] > Sent: Saturday, June 11, 2005 11:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Voicemail and MS Exchange > Synchronization > > > On 6/10/05, Dean Collins <[EMAIL PROTECTED]> wrote: > > Actually I think that has changed to 75gb now (or about to change). > > > Really? any links to support that? Since when is Micro$oft so easy on > giving up on licensing fees? I'm curious, too. If this is true it might save us a lot of pain, upgrade wise. We've been looking at moving away from Exchange entirely because of that damn 16-gig limit, and Exchange Enterprise Edition is just too expensive. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Voicemail and MS Exchange Synchronizatio n
> -Original Message- > From: Iassen Hristov [mailto:[EMAIL PROTECTED] > Does this matter? All we are saying is that Exchange supports > IMAP and we > would use IMAP as the protocol to delete the message from the user's > mailbox. How does the user access his mailbox is his choice. I think two threads of discussion got crossed. Somewhere along the line someone brought up the idea of having Asterisk act like an IMAP *server* where people could retrieve their voicemails. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to support trunking .... without zaptel timing
When I start Asterisk, I receive these errors: Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'zoom_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'zoom_trunk' without zaptel timing I have a TE110P card and IAX setup between two servers that appear to work fine. The only reason this has come to my attention is when I was trouble shooting an error with our Mitle SX200 that was reporting: T1/BRI Card at 02 06 00 00 Has exceeded the maint loss frame threshold. My main question is in regards to the timing error but if anyone has any Mitel experience (scott w!) that has seen this other error.= I'd be happy to hear about it Here is zapata.conf: [trunkgroups] [channels] musiconhold=default busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes callgroup=1 pickupgroup=1 immediate=no signalling=em_w context=zap-incoming group = 1 channel => 1-17 group = 2 channel => 21-24 Here is zaptel.conf loadzone= us defaultzone = us span=1,1,0,d4,ami e&m=1-24 Here is lsmod: Module Size Used by snd_pcm_oss47648 0 snd_pcm83336 1 snd_pcm_oss snd_timer 23812 1 snd_pcm snd_page_alloc 9604 1 snd_pcm snd_mixer_oss 16896 1 snd_pcm_oss snd51044 4 snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 10080 1 snd ipv6 232320 10 wcte11xp 25760 21 zaptel224132 43 wcte11xp i2c_i8018204 0 i2c_core 21392 1 i2c_i801 hisax 483920 0 crc_ccitt 2176 2 zaptel,hisax isdn 128716 1 hisax slhc6912 1 isdn ext3 124424 4 jbd55064 1 ext3 genrtc 9608 0 evdev 9088 0 pcspkr 3940 0 parport_pc 33220 0 parport33864 1 parport_pc piix9988 0 [permanent] ehci_hcd 30728 0 pci_hotplug31152 0 uhci_hcd 29584 0 usbcore 107896 3 ehci_hcd,uhci_hcd tg379364 0 ide_generic 1408 0 [permanent] ide_cd 38020 0 ide_core 115668 3 piix,ide_generic,ide_cd cdrom 36384 1 ide_cd font8448 0 ata_piix9092 10 libata 42756 1 ata_piix unix 26804 12 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 connecting remote villages in westernUganda
Mark, This is a wonderful thing to do for underserved societies like Uganda. The datasheet you have provided and the layout could be the model for many other developing societies both In Africa as well as central and South America. Kudos to Inveneo.org under your able leadership. Keep up the good work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Summer Sent: Monday, June 13, 2005 1:56 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk connecting remote villages in westernUganda Hi, I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa. The system is up and running since this past Wednesday (June 8th). We have installed 5 units, 4 of which are in villages with with no access to power. The system provides Internet access and phone capabilities to the users. Phone calls among the connected villages are free of charge, with the ability to place and receive calls to / from the Ugandan phone network and voice mail boxes for each station. The systems are linked using 802.11 WiFi links. For more information please have a look at the following links: For more detailed information and pictures of the Uganda deployment: http://www.inveneo.org/?q=uganda For more information about the solution we have built and implemented, here is a link to our PDF datasheet: http://www.inveneo.org/download/inveneoDatasheet.pdf And of course our website: http://www.inveneo.org/ Thank you! Mark Mark Summer co-founder, Inveneo web: http://www.inveneo.org phone: +1-415-901-1969 x 1200 FWD: 603303 cell: +1-415-867-9751 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Digium Website Update: Asterisk Business Edition
http://www.0xdecafbad.com";> > Protecting freedoms by putting limits on (thus restricting freedoms). > Interesting concept. It maybe an interesting concept, but it is absolutely true. True anarchy (no rules what so ever) cannot exist. Your freedom to kill me would impose on my freedom to live. Lift all laws and the law of the universe seems to come into play. The strong rules the weak. You end up with a dictatorship. To keep something free, there must be a law stopping it from not becoming not free. (bad english, but there it is. :) ) -- And, did Guloka think the Ulus were too ugly to save? -Centauri ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID in AMP with 2+ incoming lines
Hello, I know that I can have DID on a single line, but will AMP support 2+ lines with DID? Has anyone tried this? Straight forward? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hiss patch
In bug 0002863 a patch is mentioned that sends hiss every 20 seconds, does anyone know who wrote this or where it is available at? Scott England ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] T1 multiplexer (or ?) for failover in large installation
Use Adtran Atlas 800. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of qrss Sent: June 13, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation The box that you are talking about sounds a lot like a DACS. You might google around on that term to see if any might have automatic failover. A DACS can be reconfigured to cross-connect various DS0s on the fly - although, no matter how fast the switchover, the carrier will always "see" that something happened. Depending upon what type of signalling you are using, the trunks could end up out of service for a period longer than it takes to switch to the backup server. Also, the system can potentially fail in several ways. At the T1 level, at the trunk level and at the application level. Depending upon the nature of the failure, the one-box does it all solution seems unlikely to work - at least not by itself. -Original Message- From: Mike Sent: Mon, June 13, 2005 11:35 am Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some > type of box (multiplexer?), then be able to plug 7 asterisk servers into that > box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has > happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connecting remote villages in western Uganda
Hi, I though some of you on this list might be interested in what Inveneo is doing in Uganda. We are a San Francisco based non-profit organization that builds rugged, low-cost, highly reliable and open- source communications systems for under-served communities around the world. We have just completed our first installation in western Uganda, Africa. The system is up and running since this past Wednesday (June 8th). We have installed 5 units, 4 of which are in villages with with no access to power. The system provides Internet access and phone capabilities to the users. Phone calls among the connected villages are free of charge, with the ability to place and receive calls to / from the Ugandan phone network and voice mail boxes for each station. The systems are linked using 802.11 WiFi links. For more information please have a look at the following links: For more detailed information and pictures of the Uganda deployment: http://www.inveneo.org/?q=uganda For more information about the solution we have built and implemented, here is a link to our PDF datasheet: http://www.inveneo.org/download/inveneoDatasheet.pdf And of course our website: http://www.inveneo.org/ Thank you! Mark Mark Summer co-founder, Inveneo web: http://www.inveneo.org phone: +1-415-901-1969 x 1200 FWD: 603303 cell: +1-415-867-9751 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Digium Website Update: Asterisk Business Edition
On Mon, 2005-06-13 at 18:20 +0200, Esben Stien wrote: > "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> writes: > > > Protecting freedoms by putting limits on (thus restricting > > freedoms). Interesting concept. > > I need to repeat here. The gpl's purpose is to protect the freedoms > that comes with free software. So, you have only the freedoms that > comes with free software as defined by the FSF. You are not allowed to > do what you like. You are constrained to the freedoms that follows the > software and I think that is a very interesting concept. > Its not a freedom if its a limit. That is my point. The GPL doesnt give freedoms it takes them away by putting limits on other peoples code, not the original authors. Now if the author is ok with infringing on the rights of others then the GPL is a good choice, however if the original author is truely fore freedom in the code process, not the double speak freedom that FSF talks about (where freedom means taking away abilities) then they should not follow like sheep and repeat what the FSF says (which on its face is an outright lie since its not freedom that it grants). > > copyright and license to use are different. > > I never claimed otherwise. If you were the person that was quoting the FSF as fact then you did. Too bad it got cut out, but you can always go back to the original post that was claiming freedom means putting limits on people other than yourself. > > > You can technically put software out there with no copyright but > > under the gpl license > > Then there would be no one to enforce the license, which would be bad. > Why do you cut out what I said when I addressed that point? I am begining to think that you are doing it intentionally now. > > it only restricts *their* code (ie modifications). > > Yes, but we also want all modifications to be free > 'we' or you specifically? We is quite a loaded word. The FSF makes a false claim that it *protects* freedoms, when all it does is limit the freedoms of others to write code. Specifically if I take a program and modify it, the original is still under whatever license I got it in, but *my* code, the modifications are MINE not the original authors. The original author has NO right to claim that it is their work, nor do they have copyright on *my* code. But by releasing it under a GPL they can force me to use a license that I may not agree with. This is the reason that I dont contribute to GPL products, I dont like the idea of someone else dictating to me how I will distribute *my* code. The default GPL makes it a lciense violation to run GPL code on a commercial (or even BSD) system. Extra stuff has to be put into the GPL license to say 'its ok if you link this against non GPL libraries and such'. That is not the default, so technically unless someone did that putting a stock GPL license has other limitations on its mere use. At least historically libc on aix, hpux, sunos (4/5), irix were all not GPL libc (I dont know with solaris now they added a bunch of gpl stuff at one point). If any of the GPL licensed software did not take an overt action to say its ok to run it on those operating systems then its a license violation. That level of selective enforcement also calls into question the legal standing of the license (if certain sections are not enforced the whole agreement can be voided on first court challenge). http://www.gnu.org/licenses/gpl-faq.html#TOCLinkingOverControlledInterface for linking proprietary code to libraries - overt actions required to make it work right http://www.gnu.org/licenses/gpl-faq.html#TOCGPLCommercially for reading up on how the license affects others who write code later http://www.gnu.org/licenses/gpl-faq.html#TOCGPLIncompatibleAlone for reading on how you cant really link against libc on a commercial operating system (or anything with a license that is not compatible with the GPL, which BSD isnt becuase it allows someone to take it, write *their own* code in addition to it and not give *their own* code out. Thus by default you cant run GPL software on a BSD licensed system, nor any commercial system *unless* the developer took an overt action to say this is ok (default GPL it is not ok). http://www.gnu.org/licenses/gpl-faq.html#TOCDistributeWithSourceOnInternet for the lack of personal privacy that the GPL forces on those that choose to release under it, specifically you *must* (section 3) provide a mailing address. If you value your privacy and dont want everyone to have your address you must pay extra to get a po box so that you, as the author of the software, can comply with section 3 of the GPL - providing copies by mailorder on physical media. This is *required* not optional. The list goes on... > > the cost of restricting freedoms on others and what they can do > > with their code > > Yes, by using the GPL you restrict everyone to the four freedoms > defined in the free software definition. This is
Re: [Asterisk-Users] Zaptel modules
as the error says did you forget how to configure FXO and FXS interfaces signalling? post your configs if in doubt... best regards On 6/13/05, Bouchra Benyelloul <[EMAIL PROTECTED]> wrote: > I can't load module wcfxs or wcfxo with modprobe command, I don't have > any error message, and when i try to start zaptel I've the error below > when : > ZT_CHANCONFIG failed on channel 4: Invalid argument (22) > Did you forget that FXS interfaces are configured with FXO signalling > and that FXO interfaces use FXS signalling? > > Anyone can help me? > Thanks. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel modules
I can't load module wcfxs or wcfxo with modprobe command, I don't have any error message, and when i try to start zaptel I've the error below when : ZT_CHANCONFIG failed on channel 4: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Anyone can help me? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Re: Digium Website Update: Asterisk Business Edition
Andrew Kohlsmith <[EMAIL PROTECTED]> writes: > ABE is a VERY SPECIFIC version of HEAD (or is it STABLE?) with > features CUT OUT and nothing added that isn't in HEAD already. This is what I mean with a custom set of features. I never claimed anything was added. > I totally fail to see the problem here. The problem arises in two specific areas. Digium needs to hold the copyright of the entire core code base to be able to use a different license. This means that we cannot depend on f.ex sndfile or any other gpl project to do a specific job. Code reuse becomes a thing you cannot take advantage of. Digium cannot ship a proprietary product that includes gpl code and this means that we have to do a lot more work instead of using proven stable free code in the core of asterisk. The other problem is the issue that free software developers are mostly (in my experience) not happy with the fact that their code would be used in proprietary software. It conflicts with the whole religion of free software. This means that fewer contributions would be expected and the development process goes slower. I can only speak for myself, but please understand the clear conflict with the whole philosophy of free software. > This is exactly what they are doing. They are supporting a very > specific branch with an eye for stability and repeatability. Yes, and as I tried to say; offer support on a said set of features. It can also be a shape asterisk must be in, but it doesn't have to be non free. > [..] Oh, it's CVS HEAD from 20050612 and anyway... what? oh, [..] > libc? [..] Six what? [..]. Digium's avoiding all this bullshit. > It's a specific version of Asterisk compiled by them. This is a > good thing, not a bad thing. They can still do this with free software. You can choose to offer support on what you want, pre compiled versions or not, but this whole idea of dual licensing is hurting us, in my opinion. -- Esben Stien is [EMAIL PROTECTED] s a http://www. s tn m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@n n ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] More on the IAD connection
AdTran can come in either flavor depending on the modules they install. It can dump analog lines or it can be fully digital and split off voice T. I would recommend the digital domain for sure. Get yourself a Digium T1 card and keep everything digital. Get a block of DIDs (20 is the norm for XO) and you will have a great solution. Don't bother with analog lines. They work but consider it this way. T1 card = around $600 and can support up to 23 voice channels with 1 control data channel. 4 Port Analog - $330 - will never support more than 4 ports and IRQ becomes issue as you add more cards Obviously with a split T, you are not coing to use all 23 channels of voice. However, you will have room to grow, get better features, and avoid IRQ problems (probably). That is my $0.02. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corwin Nichols Sent: Monday, June 13, 2005 9:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] More on the IAD connection As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least workable? Thanks, -Corwin Nichols ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Cepstral partnership with Digium
You will be able to purchase Cepstral voices from Digium just like you dor for G729 already. I would guess it's 1 way to show the power of asterisk by putting all the TTS orders thru a company such as Digium. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More on the IAD connection
As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least workable? Thanks, -Corwin Nichols ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] snom 190: dial tone without registration?
Gavin Hamill <[EMAIL PROTECTED]> wrote: > On Monday 13 June 2005 16:42, alan wrote: > > > I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use > > in an Asterisk PBX/call center environment. enforce SIP channel IP restrictions in Asterisk without "host=", or get the snom 190 to stop complaining when it's not registered > How about tackling this with iptables and matching specific IP addresses on > specific MAC addresses? This solves part, but not all, of the problem. This ensures that only authorized devices can connect to asterisk, and that their IP addresses are also correct. But it doesn't force each device to use only its assigned sip channel. (That is: with dynamic IP registration, a valid IP/MAC could be configured with another device's SIP registration information, and steal calls which should be going to the other device.) I suppose iptables in combination with sip secrets should be enough. But realistically, I can already do what I want the way I want to do it, with the SPA-841. I mostly need to decide: if this feature is lacking, is it enough for me to prefer the Sipura over the snom? Thanks again, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] T1 multiplexer (or ?) for failover in large installation
The box that you are talking about sounds a lot like a DACS. You might google around on that term to see if any might have automatic failover. A DACS can be reconfigured to cross-connect various DS0s on the fly - although, no matter how fast the switchover, the carrier will always "see" that something happened. Depending upon what type of signalling you are using, the trunks could end up out of service for a period longer than it takes to switch to the backup server. Also, the system can potentially fail in several ways. At the T1 level, at the trunk level and at the application level. Depending upon the nature of the failure, the one-box does it all solution seems unlikely to work - at least not by itself. -Original Message- From: Mike Sent: Mon, June 13, 2005 11:35 am Hi, Please forgive my terminology, still a bit new to T1s and such. I'm looking for a way to have 5 T1s from a carrier terminate into some > type of box (multiplexer?), then be able to plug 7 asterisk servers into that > box (each with single port T1 card) and be able to have 2 * servers go down at any given time and not actually have the carrier see that anything has > happened. Obviously if a * server crashes the calls on it at the time will drop, but then once the box (multiplexer?) sees that a T1 is down (between the box and asterisk) it will terminate those DS0's on another T1. Basically some type of hunting/pooling/load balancing. Anyone heard of anything like this? Or am I off my rocker? Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
On Monday 13 June 2005 12:38, The VoIP Connection wrote: > This is a very interesting converation, but it seems like the BIZ forum > might be more appropriate... How on earth is this a business-related discussion? -dev would have been my guess. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 code
On Monday 13 June 2005 12:06, Matt Riddell wrote: > Race Vanderdecken wrote: > > Also subscribe to the asterisk-dev mail list. Watch it for a couple of > > days before you ask a question or they will eat your lunch. > > Or even more likely, eat you for lunch! > > :D Phew! I thought lunches was going to start disappearing... -- Steve Szmidt "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] DNIS and DID seeking confirmation
DID number is the number commonly assigned to a PSTN trunk. DNIS and DID may be the same. DNIS refers to the "Dialed Number" that is passed as signaling with the call (or on ss7). Most calls have ANI and DNIS. Your extensions look ok, assuming that the carrier sends the digits that match. What Asterisk "looks for" is determined by how you have signaling setup in your config for the card(s) that you have installed. So, this must match the signaling on the carrier side. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Millican Sent: Monday, June 13, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DNIS and DID seeking confirmation Hello all, After much googling I have come to the conclusion that in asterisk land DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are used rather interchangeably. If this is an incorrect assumption Please correct me. Based on this assumption if I have everthing set up to land in the [incoming] context and an 800# such as 1-800-123-4567 with 4 digit DNIS I can have an entry in my incoming context exten => _4567, 1, do something this is where the call to my 800 number will land regardless of which trunk the call comes in on. Like wise if I have a DID number 456-7891 with an exten=> _7891,1,do something else this will also work. Is this correct or am I way off base? Also what is Asterisk looking for as far as a delimiter or is that in a config file? Something like Seize (Wink) DNIS (Wink) ANI (Wink) Answer or Seize (*) DNIS (*) ANI (*) Answer John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition
This is a very interesting converation, but it seems like the BIZ forum might be more appropriate... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] > -Original Message- > From: Lee Howard [mailto:[EMAIL PROTECTED] > Sent: Monday, June 13, 2005 11:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Asterisk forking, Was: > Digium Website Update:Asterisk Business Edition > > Andrew Kohlsmith wrote: > > >On Saturday 11 June 2005 19:51, Lee Howard wrote: > > > > > >>I don't think that "lack of mindshare" completely defines > the reasons > >>behind Asterisk fork failures. It places all of the blame on the > >>forkers. I think the truth, though, is that they not only > fail due to > >>"lack of mindshare" but also due to competition from Digium's own > >>Asterisk community. Forks are not succeeding, yes, but > Digium has a > >>hand in that... of course they do. > >> > >> > > > >I'm not saying you're wrong, but I'm curious: how does Digium have a > >hand in a fork failing? > > > > > > That's what I tried to explain in my last post, in particular > after this first statement. Forks enter a "hostile > competition" rather than a "healthy competition". > > >>I've heard more talk about Asterisk forks than I've ever > heard about > >>forks of any other other open-source project. I think that > this says > >>something about how difficult-to-swallow Digium's > dual-license decree > >>is for a lot of prospective contributors/developers. > >> > >> > > > >I disagree; if it were that hard to swallow the project > would either be > >90% digium-written (it's not) or it would be a total flop > (again it's not). > > > > If you (or someone else reading this post) is in a position > to give statistics on what percentage of the code is > Digium-written (or Digium-rewritten - in the case where a > disclaimer is not obtained for some unpatented work and > Digium rewrites the work independently) then I would be > thrilled to see it. > > >>We see this happen all of the time with the Linux kernel. > It happens > >>with HylaFAX. It happened with X. I'm sure it happens a lot with > >>many other open-source software projects. It happens easily and > >>usually is a "healthy" process because the playing field is even. > >> > >> > > > >Agreed. But where are the successful Asterisk forks? > > > > > > I don't know of any successful Asterisk forks (unless > http://www.asteriskwin32.com is considered "successful" - > although I'll admit that I'm not really in-the-know). But > this was my point: that the way things were set up by Digium > makes a successful fork difficult. > Digium always has an upper-hand, and things were set up > intentionally this way. Again, I don't take particular issue > with this. I'm just trying to explain why forking Asterisk > would not be a particularly easy task. > > >>Of course, this "healthy" forking cannot be done with > Asterisk because > >>Digium will not accept any non-disclaimed code into their > repository. > >> > >> > > > >... What you'd described about distribution-maintained patches has > >nothing to do with this. Digium could take a > distribution-maintained > >patch and rewrite it into Asterisk proper under the dual license (as > >could any other > >contributor) and you'd still gain the benefit of the patch. I'm not > >sure I see where you're going here. > > > > > > If you (or someone else reading this) has the necessary > information to provide statistics on how what percentage of > the code comes from rewrites of non-disclaimed code, then I > would be particularly interested in hearing it. I suspect, > though, that it is a rather small - perhaps insignificant - > amount. But, yes, providing that there is not a patent > involved - yes, the work could be rewritten and integrated. > But this was my point: that given the right environment forks > can benefit from each other. > > The one thing that an Asterisk fork can never do, though, is > relicense itself. Only Diguim can do that. If Digium had > wanted an equal footing in this regard then Asterisk would be > LGPL or BSD or something a bit more liberal. So if I'm a > manufacturer of PBXes and have some proprietary IP that I do > not wish to be GPLed, then if I want to use Asterisk somehow, > then I can really only work with Digium for licensing. All > of the other forks will be license-prohibitive. > > >I have to admit that I know quite a few people with their > own modules > >and such to replace what they feel is bad code and just won't > >contribute it back to Asterisk due to the friction they've received > >about the patch. I, on the other hand, tend to bitch loud and > >continuously enough and wear them down to the point of > accepting it. > >:-) > > > > > > So we're not in disagreement, it would seem. Getti
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
"trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> writes: > Protecting freedoms by putting limits on (thus restricting > freedoms). Interesting concept. I need to repeat here. The gpl's purpose is to protect the freedoms that comes with free software. So, you have only the freedoms that comes with free software as defined by the FSF. You are not allowed to do what you like. You are constrained to the freedoms that follows the software and I think that is a very interesting concept. > copyright and license to use are different. I never claimed otherwise. > You can technically put software out there with no copyright but > under the gpl license Then there would be no one to enforce the license, which would be bad. > it only restricts *their* code (ie modifications). Yes, but we also want all modifications to be free > the cost of restricting freedoms on others and what they can do > with their code Yes, by using the GPL you restrict everyone to the four freedoms defined in the free software definition. This is exactly what we want. > The BSD license for example lets your code remain free while giving > people the freedom to create code of their own, as a modification of > yours, and use their code how they want. This is exactly the reason I choose GPL, because it doesn't allow people to do whatever they want. They only have the freedoms that comes with free software, which is exactly what we want. This ensures that the code stays free and any modification too it is also free. This is what we want and you obviously want something else. When we, the saints of the church of emacs, speaks about free software we are referring to the freedoms that comes with free software (nothing more, nothing less). Free software has a definite definition for us, which is that of the fsf. > If people want your version they can always get that from you, and > so it is intact as 'free'. Yes, but we also want the modifications to the software to be free. We basically want what's defined in the GPL. > It does not give full unrestricted modification clauses. You can modify it as much as you want as long as the modifications also are free, just as the original code. > proposed GPL 3.0 I rather not discuss GPL 3.0 before a draft. > Your version which you released 'free' would still be there. In its > unmodified glory. By using the GPL, we also ensure that any modification to it, be free. This is desired. > The GPL does not ensure freedom to all It ensures the freedoms that are defined in the free software definition. > it works like a parasite and infects future code Yes, this parasitic effect is exactly what we want. > All it does is force others who write code to be assimilated into the > same doctrine. Yes, which is exactly what we want. If you choose to use GPL code, you have to follow the rules. > I guess what I am trying to say is that GPL does little to protect the > original author The copyright protects the original author by law. > it removes freedoms from subsequent authors by forcing them to > license in the same way. Yes, and that's what I love about free software. The software stays free. > it doesnt guarantee the freedom of subsequent authors, it curtails > that freedom. Once again, it only guarantee freedoms that follow free software. > And you can copyright (and infact do) without the GPL. Yes, but we use the gpl to protect the freedoms that follows free software. > The GPL is *not* a copyright it is a license for use. They are very > different things. You can copyright something and distro it without > GPLing it. Indeed. > The free software continues to be as free as the author wants. Yes, the copyright holder can do whatever he feels like with the code. Once he puts a GPL on it and release it, the code is free for ever and any modifications to it is also free. By holding the copyright, he can also choose to change the license, but only on the code that he holds the copyright of. The code that was released as free, however, stays free. > it does however curtail the freedoms of any subsequent authors that > enhance the code. Which again, it's the desired effect. > subsequent authors now have *no* choice in how they license it, they > are forced to license it the same way as you, which curtails > freedom. Yes, glad you understand cause this is the purpose. The freedoms that follow free software will continue to follow it and neither you nor anyone else can change that. > The modifications are the *only* difference between what you release and > what they release, so if they use your code as a base and make changes > to suit a particular need, their code, which they did write all of, > cannot be licensed how they choose This is exactly what we want. > the parasitic nature of the GPL means that their modifications, > *their* code, must also be GPLed You're just explaining what we want. > The GPL doesnt protect freedom, it curtails freedom of fu
RE: [Asterisk-Users] Interfacing to an IAD
I have had experience with both the Vina and with XO. If you ask for it, you should be able to get an Adtran 600 series on the circuit. I never had any success with the Vina and it really is not a piece of equipment I would bet the farm on. They may have improved but I would still just as fo ran AdTran since it definitely has a T1 interface that works. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corwin Nichols Sent: Monday, June 13, 2005 8:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Interfacing to an IAD I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an "IAD" which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a "Lucent Digital Vina Integrator" IAD with a digital TC card. I've searched the web to find any sort of info on how I can connect this IAD to my Asterisk box without success. What I find in general is that this kind of IAD can either provide analog voice output (POTS) or "digital T-1" output. I presume the latter is what they're providing. Is a "digital T-1" from an IAD the same kind of interface as a PRI T-1? Would something like a Digium TE-110P handle this interface? Does anyone out there have experience with a Vina IAD with digital voice circuit output? Thanks for any assistance. -Corwin Nichols ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] 407 Proxy Authentication Required
Title: 407 Proxy Authentication Required We also have the same problem over long latency networks – ATA also gives Call Rejected: 407. We have tried a lot of different phones and soft phones and the only one working is Xten. In any case this is apparently only problem with newer versions of * - you can use very old version you can avoided the problem. We were not yet able to find final solution for this problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahan Kalutanthri Sent: Monday, June 13, 2005 3:20 AM To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] 407 Proxy Authentication Required I am getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network using sjphone & snom. How to overcome this..!! Pls advice..! Shahan This e-mail may contain confidential and/or privileged information. If you are not the intended recipient or have received this e-mail in error, please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Trunking
I'm trying to setup SIP trunking between 2 asterisk servers. Eventually
there may be up to 5 servers linked together depending on the growth
needed. I have IAX2 trunking working, but I want both.
For simplicity, I have named the two servers, alpha and beta. Extension
7100 is a Polycom IP600 on alpha and extension 7300 is using kphone on
beta. Both ae using SIP. Below are the relvenant parts of
extensions.conf and sip.conf.
;# server alpha
; extensions.conf
[staff]
extension => 7100,Dial(SIP/7100)
extension => 7300,Goto(siptrunk,7300,1)
[siptrunk]
include => siptrunk-beta
[siptrunk-beta]
exten => _73XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=
;# server beta
; extensions.conf
[staff]
extension => 7300,Dial(SIP/7300)
extension => 7100,Goto(siptrunk,7100,1)
[siptrunk]
include => siptrunk-alpha
[siptrunk-alpha]
exten => _71XX,1,Dial(SIP/siptrunk-peer/${EXTEN})
; in sip.conf
[siptrunk-peer]
type=peer
username=siptrunk-peer
secret=password
host=
[siptrunk-user]
type=user
username=siptrunk-user
secret=password
host=
When dialing 7300 from alpha, I get the following:
-- Executing Dial("SIP/7000-e924", "SIP/siptrunk-peer/7300") in new
stack
-- Called siptrunk-peer/7300
Jun 13 11:10:47 WARNING[18099]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 45a607bc5f66cfb363f2cc565b85fa29@address> for seqno 102 (Critical Request)
== No one is available to answer at this time
Any ideas?
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