Re: [Asterisk-Users] Asterisk and Cisco Call Manager
try with
type=peer
good luck
Edgar
On Tue, 2005-04-26 at 13:08 +0200, Alessio Focardi wrote:
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
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Re: [Asterisk-Users] Asterisk@home 0.9 and Cisco Callmanager
This is my sip trunk configuration canreinvite=yes context=from-internal disallow=g729 host=192.168.1.138 mask=255.255.255.255 qualify=yes type=peer maybe it could be the context! HTH Edgar On Thu, 2005-04-21 at 12:57 +0800, Dinesh wrote: Great:) Just one question, I am trying to get the cisco callmanager with [EMAIL PROTECTED] integration. Having some problem, if I edit the config files, the [EMAIL PROTECTED] doesn't see the sip peers. I am following this http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%20CallManager% 20Integration It is working fine on a asterisk installation. Regards, Dinesh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, April 14, 2005 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.9 released More bug fixes. *69 works now. Cisco stuff works. Lots of other fixes. A wakeup call feature was added on *62 http://asteriskathome.sourceforge.net/ Discussion Forums http://sourceforge.net/forum/?group_id=123387 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Phone book
Hello i need to make a central phone book, at this time we got a lot of offices far away from here and i want to know if it possible to get a central phone book from ldap or mysql to make calls just typeing the name of the office, i saw the macros extensions using ldap but just to get the caller id, i need to replace the extension get by an ldap field, but dont know if this can be done. TIA Edgar ___ 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 phone service for linux
Hello, i want to be able to use my zultys softphone to make calls pc-tp-pc and pc-to-phone, from my home, i want to install an asterisk server but at this time i need to connect to a voip service provider, can anybody tell my wich provider are the best and got good rates??? TIA Edgar ___ 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] Problem with Asterisk Unable to allocate channel
Hello, yesterday when i wasnt in the office the asterisk server stop working, it was registering the sip terminals but cant make calls, because im not in the office i told the people to reboot the server to make the server works again but today i found this lines in the full log, can anybody tell me what was happend? Feb 28 17:30:07 WARNING[6351]: Unable to allocate channel structure Feb 28 17:30:07 DEBUG[6351]: build_route: Contact hop: luiscarlos sip:[EMAIL PROTECTED]:5060 Feb 28 17:30:07 NOTICE[6351]: Unable to create/find channel TIA Edgar ___ 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] How to Create customized audio file to use with ASTCC??
Hello Mark, i tried to get the spanish soun but get You don't have permission to access /VoIP/AsteriskSounds_ES.tar.gz on this server. can you help me??? TIA Edgar Hi Derek, Yes there is. Take a look at my web pages http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a project to record as many different regional accents (with local lingo) as I could. It started well but as I had to rely on others to create the files (I don't speak with a Welsh accent) it fell by the wayside. If you'd like to voice your files and also know a woman whom would be willing to do the same (female voices are more sought after) then I could do the editing for you. Take a look in your asterisk-sounds directory for the script to the files. I used a Radioshack microphone and recorded the file as a WAV with Audacity then chopped it up into the phrases. If anyone else is interested in getting this project going again please contact me off list. Mark Derek Conniffe wrote: I did this - I'm in Ireland and needed sounds like Euro and Hash rather than Dollars and Pound. I typed up the script of what was needed, recorded it a number of times on semi-professional equipment and then I spent the time editing the recordings into the individual wav files and then, finally, converted the sounds into gsm files. These sounds are being used in a low cost call shop in Dublin now. I'm not sure if my ASTCC recordings would suit your (or anyones) needs but if you would like a copy I have no problem providing them publically for no charge. Derek Daniel Eboa wrote: Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesnt match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. ___ 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] howto answer a call in a queue
My implementation accepts fw calls, i use the # key to transfer the calls and works very well, im using zultys softphone, im very happy with that phone, thanx for all your help Edgar Here you find a complete working pbxconfig for asterisk. http://www.gwsnettech.de/work/astconfig.txt This config is our production config and works quite well. Perhaps there are some curious constructions in there, but they work for us ;-) With the queue functions we have only these problems: joinempty=no ; does not work leavewhenempty = yes ; does not work Incoming calls to the queue cannot be transfered with # transfer, even though the queue command has the tT options... Hope this helps Guido -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 15:34 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue On the CLI everything seems to be ok, the call enters the queue plays the message, on the CLI, appear as a call entering the queue and then show a message saying wich agent is assigned to it! can you send me your config, maybe there is something im doing wrong, thnx for all your help!! Edgar I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent
[Asterisk-Users] gsm audio files
Hello, anyone knows if exist the audio files in spanish?? or how can i record the voice in gsm extension??? can i play for some announce a random file?? TIA Edgar ___ 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] howto answer a call in a queue
Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ 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] howto answer a call in a queue
Thnx i would let you know my results!! Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ 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
RE: [Asterisk-Users] howto answer a call in a queue
I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ Asterisk-Users mailing list
RE: [Asterisk-Users] howto answer a call in a queue
On the CLI everything seems to be ok, the call enters the queue plays the message, on the CLI, appear as a call entering the queue and then show a message saying wich agent is assigned to it! can you send me your config, maybe there is something im doing wrong, thnx for all your help!! Edgar I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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
RE: [Asterisk-Users] howto answer a call in a queue
im using zultys sw phone, very nice and works very stable!! TIA Edgar I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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] howto answer a call in a queue
hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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] howto answer a call in a queue
Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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] Call queue ackcall doesnt work
Hello, i got configured the queues.conf and agents.conf and works well in the first configuration for testing purposes i used [agents] autologoff=15 wrapuptime=5000 ackcall=no group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera and when i loged in, plays a musiconhold, and when a call enters the queue rings once and the call is conected directly, later i use the ackcall=yes options, to got to press # key to answer the call, but only ring once and then connect directly, how can i configure this to ensuer the agent press # to answer the call?? TIA Edgar ___ 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] Group Extension
Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? TIA Edgar ___ 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] Callmanager and Asterisk problem
Hello everybody, i got and asterisk and a CCM configured thru SIP, and in the sip show peers appear Name/usernameHostDyn Nat ACL Mask Port Status CCM 10.60.27.138255.255.255.255 5060 OK (1 ms) but when i enabled sip debug in the CLI got this Sip read: SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK784b4a8c From: asterisk sip:[EMAIL PROTECTED];tag=as7b541ffe To: sip:10.60.27.138 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.60.27.138 SIP/2.0 Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK4aaa1423 From: asterisk sip:[EMAIL PROTECTED];tag=as6f4153c7 To: sip:10.60.27.138 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 26 Jan 2005 09:15:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.60.27.138:5060 can anybody help me?, what could be the problem?? when i try to call an ccm extension got the busy signal, TIA Edgar ___ 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] Guatemala DID's?
Hello Phil im from Guatemala, im living in Madrid but im thinking in came back in july, if its helps to you, im thinking in make an installation of asterisk to make calls, if you found something now to make calls please inform me! TIA Edgar I'm looking for a company that offers Guatemala DID's. I saw that Lingo does, but Lingo isn't easily compatible w/ Asterisk, so they're a last resort. Thanks in advanced, Phil Astin. ___ 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] X-lite for linux
Hello im triying to config xlite on wine for linux, but got problems with the mic test, can anybody tell me how to get the mic config to work with wine or x-lite? TIA Edgar ___ 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] X-lite for linux
Thanks, its a better solution!!! Edgar Hi Edgar, Don't use XLite under wine, use native code for Linux: http://sipthat.com/archives/000187.html Cheers Edgar de Leon escreveu: Hello im triying to config xlite on wine for linux, but got problems with the mic test, can anybody tell me how to get the mic config to work with wine or x-lite? TIA Edgar ___ 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 -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 Project Manager Devel-IT - http://www.devel.it TDKOM Group ___ 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] Callmanager 4.1 and Asterisk
i apreciatte if u can send me the conf files, and the screenshots about the CM config, its really easy as you said, i like asterisk very much, after that we are planning to make test on echo and relay calls, but think it would work great, thanx for your help, Edgar You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk) In the trunk configuration change the transport to UDP. Enter the IP of Asterisk. And create a route pattern with gateway the SIP trunk In Asterisk in extensions.conf create the route to CCM phones. I have this setup in my lab with CCM 4.02sr1 and works so fine. If you need the sip.conf / extensions.conf and an screenshot of the route pattern and SIP trunk config just let me know! Happy holidays! Keith O'Brien [EMAIL PROTECTED] wrote: I have a similar setup. To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk. Keep the physical phones registered to CM. From there setup a dialplan on both Call Manager and As terisk to relay calls between the two systems. For example, assign all physical phones extension 2XXX and softphones 3XXX. Have asterisk route 2XXX calls to CM via SIP and vice versa on Call Manager. Also, just so that you are aware you can register a SIP Linux softclient to Cisco Call Manager if you are running Version 4.1 --- Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control?? or if i need to declare all the extensions in the asterisk?? can anybody help me?? TIA Edgar ___ 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 - Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more.___ 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
[Fwd: [Asterisk-Users] Callmanager 4.1 and asterisk]
Anyone??? Mensaje original Asunto: [Asterisk-Users] Callmanager 4.1 and asterisk De: Edgar de Leon [EMAIL PROTECTED] Fecha: Mar, 28 de Diciembre de 2004, 8:21 am Para: asterisk-users@lists.digium.com -- Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control?? or if i need to declare all the extensions in the asterisk?? can anybody help me?? TIA Edgar ___ 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] Callmanager 4.1 and asterisk
Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control?? or if i need to declare all the extensions in the asterisk?? can anybody help me?? TIA Edgar ___ 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
