Re: [Asterisk-Users] Extensions Problem
> So, I assume we need to implement 9, and the number. However, when I
> do this, the 9 gets passed on to our SIP provider, which tries to dial
> 9NXX, and all goes to hell.
>
> Question - is there a way to allow 9 in the dialing plan, without having
> it be passed to the sip provider.
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
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RE: [Asterisk-Users] Extensions Problem
Phillip, exten => _9NX,1,StripMSD,1 Exten => _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten => _NX,2,Congestion Should work Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Phillip Jackson Sent: 26 October 2003 23:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Extensions Problem Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten => _NX,2,Congestion Exten => _1NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten => _1NX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extensions Problem
You may have a file called dialplan.xml being TFTP'd to your phone. It has a number of rules in it for helping the phone to determine when it has complete number. It may need some tuning to bring it in line with what you need. I've found that the phone appears to treat the contents of the file as a hash rather than as a sorted list. That is, certain rules that appear later in the file actually get used before rules earlier in the file. I think the rules get used in a 'shortest match first' scenario. Regards, Ray Burkholder http://www.oneunified.net 704 576 5101 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Phillip Jackson > Sent: October 26, 2003 18:35 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Extensions Problem > > > Hello again, > > Here's the next big issue, I thought I'd let you munch on. > We are utilizing > Cisco 7960's and the following entries in our extensions.conf file: > > Exten => 1637,1,Dial(SIP/100) > Exten => _NX,1,Dial(SIP/[EMAIL PROTECTED]) > Exten => _NX,2,Congestion > Exten => _1NX,1,Dial(SIP/[EMAIL PROTECTED]) > Exten => _1NX,2,Congestion > > These extensions allow us to utilize our SIP provider - ONLY > when being dialed > from a regular telephone attached to a Cisco ATA-186. Our > Cisco 7960 only > allows us to dial 4 charachters before it tries dialing. So, > I assume we need > to implement 9, and the number. However, when I do this, the > 9 gets passed on > to our SIP provider, which tries to dial 9NXX, and > all goes to hell. > > Question - is there a way to allow 9 in the dialing plan, > without having it be > passed to the sip provider. > > Regards, > Phillip > > > -- > Phillip C. Jackson > [EMAIL PROTECTED] > > - > This mail sent through IMP: http://horde.org/imp/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Scanned for viruses and dangerous content at > http://www.oneunified.net and is believed to be clean. > -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] extensions problem
Maybe try, [dialjon] exten => s,1,answer exten => s,2,Dial(SIP/2000,15) exten => s,3,congestion exten => s,4,Playback(noone) exten => s,103,Goto(onphone,s,1) Not sure if it will work.. just thinking, -Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence Sent: Monday, March 15, 2004 10:29 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] extensions problem Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include => dialjon [inboundB] include => dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten => s,1,answer exten => s,2,Dial(SIP/2000,15) exten => s,3,Playback(noone) exten => s,103,Goto(onphone,s,1) Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. Can anyone point out where I've gone wrong ? TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] extensions problem
Your phone supports call waiting, so isn't giving out busy. I had the same problem with a budgetone 102, you can't turn this off on the phone but you can work round it by adding Incominglimit=1 Into the sip.conf entry for the phone >From: Jon Lawrence <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Date: Mon, 15 Mar 2004 15:29:01 + >Subject: [Asterisk-Users] extensions problem >Reply-To: [EMAIL PROTECTED] >Hi, >I've got 2 x100p's installed in my system. >Both execute the same incoming contexts as follows: >[inboundA] >include => dialjon >[inboundB] >include => dialjon|09:00-16:30|Mon-Fri|*|* > >[dialjon] >exten => s,1,answer >exten => s,2,Dial(SIP/2000,15) >exten => s,3,Playback(noone) >exten => s,103,Goto(onphone,s,1) > >Am I right in saying: >if no one answers at ext 2000 then s,3 is executed. >if ext 2000 is busy then 103 is executed. >If so then sometihng is wrong. If I'm already on a call, I want 103 to be >executed however, this isn't happening. If a new call comes in (whilst I'm >talking on ext 2000) I here it ringing on my handset. >Can anyone point out where I've gone wrong ? >TIA >Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem
On Monday 15 March 2004 16:26, Asterisk DEV. Mailing List wrote: > Your phone supports call waiting, so isn't giving out busy. I had the > same problem with a budgetone 102, you can't turn this off on the phone > but you can work round it by adding > > Incominglimit=1 > > Into the sip.conf entry for the phone I can imagine situations where call waiting might be useful, but only if I can acknowledge the call with the phone either rejecting it to a queue or ditching the current call and picking up the incoming one - something to play with in the future (once I've found a way of getting UK callerID working). I've added the Incominglimit=1 and that's fixed my immediate problem. Thanks everyone. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
Jon Lawrence wrote: Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include => dialjon [inboundB] include => dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten => s,1,answer exten => s,2,Dial(SIP/2000,15) exten => s,3,Playback(noone) exten => s,103,Goto(onphone,s,1) Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. It depends on your SIP device. Asterisk places the call to your SIP device regardless, since by SIP protocol design the UA is not a "slave", it is free. So the SIP ua must answer "busy" for Asterisk to understand that you're busy. If not, the call is placed to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can limit the number of concurrent calls. There's some code in Asterisk chan_sip.c to limit the number of calls placed to a SIP phone, but right now it's not working at all. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] extensions problem (SIP)
In your SIP.conf set callwaiting = no. This will work for single registrations. If you have multiple call appearance on you phone, then it will just ring to the second line (e.g. Cisco 7960). If you only have a single registration, then you should be fine. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Monday, March 15, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] extensions problem (SIP) Jon Lawrence wrote: > Hi, > I've got 2 x100p's installed in my system. > Both execute the same incoming contexts as follows: > [inboundA] > include => dialjon > [inboundB] > include => dialjon|09:00-16:30|Mon-Fri|*|* > > [dialjon] > exten => s,1,answer > exten => s,2,Dial(SIP/2000,15) > exten => s,3,Playback(noone) > exten => s,103,Goto(onphone,s,1) > > > Am I right in saying: > if no one answers at ext 2000 then s,3 is executed. > if ext 2000 is busy then 103 is executed. > > If so then sometihng is wrong. If I'm already on a call, I want 103 to be > executed however, this isn't happening. If a new call comes in (whilst I'm > talking on ext 2000) I here it ringing on my handset. > It depends on your SIP device. Asterisk places the call to your SIP device regardless, since by SIP protocol design the UA is not a "slave", it is free. So the SIP ua must answer "busy" for Asterisk to understand that you're busy. If not, the call is placed to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can limit the number of concurrent calls. There's some code in Asterisk chan_sip.c to limit the number of calls placed to a SIP phone, but right now it's not working at all. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Monday 15 March 2004 16:00, Olle E. Johansson wrote: > > It depends on your SIP device. Asterisk places the call to your SIP device > regardless, since by SIP protocol design the UA is not a "slave", it is > free. So the SIP ua must answer "busy" for Asterisk to understand that > you're busy. If not, the call is placed to you and Asterisk has no > knowledge that you are busy. Check you SIp phone if you can limit the > number of concurrent calls. So does anyone know if the Grandstream handytone-286 sends this "busy" answer ? I'm guessing it doesn't. In that case, what other ways are there of connecting my dect phones to a voip * system ? - can I just connect them into the x100p's phone socket (how do I send calls to that port) or do I need to get a fxs card and run wire's everywhere - not an option :( How does everyone else connect up DECT phones to a * based system. Surely * should know if a phone is in use ? After all it initiated/took part in the call in the first place ;) Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
Jon Lawrence wrote: Surely * should know if a phone is in use ? After all it initiated/took part in the call in the first place ;) Again, the SIP device is not a "slave" device. It could receive a call from somewhere else and be busy without Asterisk having a clue. A lot of SIP UAs, like Xten software and the SNOM 200 phone support multiple SIP accounts. So Asterisk could keep track of how many simultaneous calls it can place to the UA, but not decide if the UA is busy or not. The outgoinglimit code is used for this, but it's disabled and non-functional right now. The incominglimit limits how many simultaneous calls a UA may place to Asterisk. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: > The incominglimit limits how many simultaneous calls a UA may place to > Asterisk. I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI then you will see the limit set. The behavior of * then will consider the device busy if there is a call in progress and the inuse count is incremented. Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 7960 and Grandstream BT-102 phones. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
Walker Haddock wrote: On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: The incominglimit limits how many simultaneous calls a UA may place to Asterisk. I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI then you will see the limit set. The behavior of * then will consider the device busy if there is a call in progress and the inuse count is incremented. Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 7960 and Grandstream BT-102 phones. The incominglimit= config is read in build_user, users place calls to asterisk. Peers have no incominglimit. Funny enough outgoinglimit= was also coded in build_user for users, even though chan_sip place calls to peers. So maybe it just happens to work as you say for "friends" that is both user and peer. The find_user routine just checks users, not peers. Would be greatful if someone cleaned up this part of chan_sip and added support for outgoinglimit for peers. Also, see bug http://bugs.digium.com/bug_view_page.php?bug_id=0001064 /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Monday 15 March 2004 20:35, Walker Haddock wrote: > On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: > > The incominglimit limits how many simultaneous calls a UA may place to > > Asterisk. > > I'm pretty sure that the incominglimit specifies how many calls that * can > send to the SIP device. If you set incominglimit=1 and then do a SIP SHOW > INUSE from the *CLI then you will see the limit set. The behavior of * > then will consider the device busy if there is a call in progress and the > inuse count is incremented. > > Paul Lieu did some work on this a few months ago and I've been using it on > my Cisco 7960 and Grandstream BT-102 phones. The interface to my handytone is identical to a BT-102 so it may also work with the handytone :). Where did you specify incominglimit=1 - is it in the sip.conf for that UA ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Mon, Mar 15, 2004 at 08:56:17PM +, Jon Lawrence wrote: > > The interface to my handytone is identical to a BT-102 so it may also work > with the handytone :). Where did you specify incominglimit=1 - is it in the > sip.conf for that UA ? Yes, put it in the stanza for the devicd. As Olle just pointed out, make sure the device is a `friend` Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Mon, Mar 15, 2004 at 09:56:04PM +0100, Olle E. Johansson wrote: > Walker Haddock wrote: > >On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: > > > >>The incominglimit limits how many simultaneous calls a UA may place to > >>Asterisk. > > > >I'm pretty sure that the incominglimit specifies how many calls that * can > >send to the SIP device. If you set incominglimit=1 and then do a SIP SHOW > >INUSE from the *CLI then you will see the limit set. The behavior of * > >then will consider the device busy if there is a call in progress and the > >inuse count is incremented. > > > >Paul Lieu did some work on this a few months ago and I've been using it on > >my Cisco 7960 and Grandstream BT-102 phones. See bug (closed), http://bugs.digium.com/bug_view_page.php?bug_id=408 This is the one that Paul Liew opened and provide a fix for. > > The incominglimit= config is read in build_user, users place calls to > asterisk. > Peers have no incominglimit. > > Funny enough outgoinglimit= was also coded in build_user for users, even > though > chan_sip place calls to peers. > > So maybe it just happens to work as you say for "friends" that is both user > and peer. Yes, I have my SIP UAs defined as `friends`. Also, like Paul said in bug 1064, you have to put the username=xxx in the stanza as well. > The find_user routine just checks users, not peers. > > Would be greatful if someone cleaned up this part of chan_sip and added > support > for outgoinglimit for peers. > > Also, see bug > http://bugs.digium.com/bug_view_page.php?bug_id=0001064 I see that Paul is commenting on this, so I don't need to get involved except to say that I am using the feature to limit the number of calls that * can make to the UA. If there is a disagreement in definition/perspective here, I guess we can change all of our sip.conf entries from incoming to outgoing easily enough. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ 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
