Re: [asterisk-users] 3-way calling
it is probably not what you are looking for. but simply use a conference room of asterisk for those 1 line phones. pamela Rilawich Ango wrote: > That's easy if phone supports 3 ways call. However, phones in my > company only have 1 line without join function. Is it possible to > implement 3 ways call using Asterisk without phone support in my case? > > On 9/28/07, Anthony Francis <[EMAIL PROTECTED]> wrote: > >> Rilawich Ango wrote: >> >>> What do you mean? I just want to know whether there is a way to do >>> the following. >>> >>> 1. A --calls --> B >>> 2. A on hold, B --calls --> C >>> 3. A, B and C connected to talk >>> >>> On 9/28/07, Paul Hales <[EMAIL PROTECTED]> wrote: >>> >>> How are you going to do it without a phone? PaulH >>> ___ >>> >>> Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ >>> >>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> your phone would need a "Join" feature or you could do it externally >> with AMI but that would be clumsy. Most Sip phones have a 3way calling >> option right on them. >> >> Anthony >> >> ___ >> >> Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ >> >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 / install app_bundle problems
Hello group, I've encountered some problems getting Beronets app_bundle installed with Asterisk 1.4.0-beta2. The output messages I get while running make are: cc -ggdb -Wall -D_GNU_SOURCE -fPIC -DAST_CONFIG_DIR=\"/usr/local/asterisk/etc/asterisk/\" -DPTYSPOOLDIR=\"/usr/local/asterisk/var/spool/asterisk/ptyspool\" -DSCRIPTPATH=\"/usr/local/asterisk/var/lib/asterisk/scripts\" -Wall -c -o app_waitfordigits.o app_waitfordigits.c app_waitfordigits.c:60: warning: type defaults to `int' in declaration of `STANDARD_LOCAL_USER' app_waitfordigits.c:60: warning: data definition has no type or storage class app_waitfordigits.c:62: warning: type defaults to `int' in declaration of `LOCAL_USER_DECL' app_waitfordigits.c:62: warning: data definition has no type or storage class app_waitfordigits.c: In function `waitfordigits_exec': app_waitfordigits.c:87: warning: initialization discards qualifiers from pointer target type app_waitfordigits.c:88: warning: initialization discards qualifiers from pointer target type app_waitfordigits.c:99: warning: implicit declaration of function `LOCAL_USER_ADD' app_waitfordigits.c:137: warning: implicit declaration of function `LOCAL_USER_REMOVE' app_waitfordigits.c: In function `unload_module': app_waitfordigits.c:196: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) app_waitfordigits.c:196: error: (Each undeclared identifier is reported only once app_waitfordigits.c:196: error: for each function it appears in.) app_waitfordigits.c: In function `usecount': app_waitfordigits.c:215: warning: implicit declaration of function `STANDARD_USECOUNT' make: *** [app_waitfordigits.o] Error 1 Is this a known issue? Or am I doing something wrong? Thanks, Pamela ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Mailbox Configuration
hello, there are two options to do this: 1. if you retrieve your voicemails via your phone you will played back some option like changing your busy and unavailable message (after dialing 0 - just follow the instructions). or 2. you just record your soundfiles with your favourite recorder and change them into *.gsm with sox (http://www.voip-info.org/wiki-Asterisk+sound+files) and place them into the directory of your voicemail. hope this helps pamela Darly Coupet wrote: Hi, How do I configure mailbox to play a different gsm prompt that default setup. I would like to replace unaivalable and busy message with a new sound file that I created. All comments are welcomed and greatly appreciated. Darly ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 passthrough
hello, i wanted to use g729 in asterisk (iax to sip) as passthrough. Has anyone got experience with configuring this or does someone know if this is possible at all? At the moment asterisk2 is always transcoding to alaw but but this results in horrible voice quality. phone1 behind NAT (sip) -> SER -> asterisk1 (iax) -> asterisk2 -> phone2 thanks for your help pamela ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom200 strange sound problem
Hello group, I've a rather strange problem with my Snom200 telephone. I'm using it in combination with SER, asterisk and rtpproxy. The telephone is behind NAT and connects to SER. It can be called without any problem from any Client on asterisk or SER. But whenever I make a call to asterisk or other clients on asterisk I have no sound until I press the Transf and Esc button in sequence. With XLite and a Snom100 I don't have these problems. They work fine in all directions, therefore I don't believe that it's a configuration issue on asterisk or ser side but I maybe wrong. I have already tried to up- and downgrade the snom200 to another firmware (current version is now snom200-SIP 2.04g) but it didn't help. Any help on this would be greatly appreciated. Pamela ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] problems with asterisk and the IAX protocol
Hi Kevin,
no you didn't miss the reply and I've not resolved it yet.
Have you got similar problems?
Pamela
Kevin Fjelsted wrote:
Pamela,
Did you resolve the problems you described?
I didn't see a reply on the list but I may have missed it.
-Kevin
-Original Message-----
From: Pamela Weis [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 05, 2004 10:22 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problems with asterisk and the IAX protocol
Hello group,
I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.
SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2
Both SER and asterisk run on a machine with a public IP address. When
the telephone on one side makes a call the telephone on the other side
rings. But whenever I pick up the call, asterisk2 hangs up without much
warning and then the telephone rings unexpectedly again and again.
Here is the output of the two asterisk machines:
asterisk 1:
*CLI>
-- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
-- Executing Dial("[EMAIL PROTECTED]/1",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/62.116.54.194-b71d is ringing
-- SIP/62.116.54.194-b71d answered [EMAIL PROTECTED]/1
== Spawn extension (local, 123, 1) exited non-zero on
'[EMAIL PROTECTED]/1'
-- Hungup '[EMAIL PROTECTED]/1'
-- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
-- Executing Dial("[EMAIL PROTECTED]/2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/62.116.54.194-6749 is ringing
---
asterisk2:
*CLI> -- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 62.116.54.194 (format G729A)
-- Format for call is G729A
-- IAX2[asterisk]/1 stopped sounds
-- IAX2[asterisk]/1 stopped sounds
-- IAX2[asterisk]/1 answered SIP/-0811bef8
Aug 5 17:57:00 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 1 (Response)
-- Hungup 'IAX2[asterisk]/1'
== Spawn extension (sip, 123, 1) exited non-zero on 'SIP/-0811bef8'
Aug 5 17:57:05 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 1 (Response)
Aug 5 17:57:06 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
-- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 62.116.54.194 (format G729A)
-- Format for call is G729A
-- IAX2[asterisk]/2 stopped sounds
-- Hungup 'IAX2[asterisk]/2'
== No one is available to answer at this time
I also have another question to asterisk and NAT:
o) If one asterisk machine and the telephones are behind NAT, do I need
a proxy to get the speech through, or should asterisk work this out on
its own?
Any help with my problem will be greatly appreciated. Thanks in advance.
Pamela Weis
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[Asterisk-Users] problems with asterisk and the IAX protocol
Hello group,
I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.
SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2
Both SER and asterisk run on a machine with a public IP address. When
the telephone on one side makes a call the telephone on the other side
rings. But whenever I pick up the call, asterisk2 hangs up without much
warning and then the telephone rings unexpectedly again and again.
Here is the output of the two asterisk machines:
asterisk 1:
*CLI>
-- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
-- Executing Dial("[EMAIL PROTECTED]/1",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/62.116.54.194-b71d is ringing
-- SIP/62.116.54.194-b71d answered [EMAIL PROTECTED]/1
== Spawn extension (local, 123, 1) exited non-zero on
'[EMAIL PROTECTED]/1'
-- Hungup '[EMAIL PROTECTED]/1'
-- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
-- Executing Dial("[EMAIL PROTECTED]/2",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/62.116.54.194-6749 is ringing
---
asterisk2:
*CLI> -- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 62.116.54.194 (format G729A)
-- Format for call is G729A
-- IAX2[asterisk]/1 stopped sounds
-- IAX2[asterisk]/1 stopped sounds
-- IAX2[asterisk]/1 answered SIP/-0811bef8
Aug 5 17:57:00 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 1 (Response)
-- Hungup 'IAX2[asterisk]/1'
== Spawn extension (sip, 123, 1) exited non-zero on 'SIP/-0811bef8'
Aug 5 17:57:05 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 1 (Response)
Aug 5 17:57:06 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
-- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 62.116.54.194 (format G729A)
-- Format for call is G729A
-- IAX2[asterisk]/2 stopped sounds
-- Hungup 'IAX2[asterisk]/2'
== No one is available to answer at this time
I also have another question to asterisk and NAT:
o) If one asterisk machine and the telephones are behind NAT, do I need
a proxy to get the speech through, or should asterisk work this out on
its own?
Any help with my problem will be greatly appreciated. Thanks in advance.
Pamela Weis
___
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[EMAIL PROTECTED]
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To UNSUBSCRIBE or update options visit:
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