[asterisk-users] Goto Queue, does not work, it should play message or any thing
Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten = 5631040,1,Playback(WelcomeMessage) exten = 5631040,2,Goto(OrangeCMG,s,1) So how I can overcome this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto Queue, does not work, it should play message or any thing
On 11/15/2011 04:56 AM, bilal ghayyad wrote: Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten = 5631040,1,Playback(WelcomeMessage) exten = 5631040,2,Goto(OrangeCMG,s,1) So how I can overcome this? Regards Bilal -- I prefer to start all my extensions with a NoOp() as priority 1. Perhaps if you try this... exten = 5631040,1,NoOp(Inbound call) exten = 5631040,2,Goto(OrangeCMG,s,1) -- The truth speaks for itself. I'm just the messenger. Lyta Alexander - Babylon 5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP endpoint registrations
Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
On 11/15/2011 07:28 AM, Faraj Khasib wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
Hi You can make the call from all. For that u need not to register but to receive the call you need to register one and that can be done by any one iphone app On Nov 15, 2011 7:03 PM, Faraj Khasib fkha...@iconnecths.com wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Frequent Asterisk Restarts
On Thu, Nov 10, 2011 at 03:02:43PM -0500, eherr wrote: Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my backtrace.txt completely useless or should I still submit? Don't bother. It makes the issue more aparent, but has a very large performance hit. In some cases it will also make the problem go away (in some odd races). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
btw the call is one direction from clients to Call center My question can be rephrased can I make call without registration to an registered SIP account? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, November 15, 2011 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple SIP endpoint registrations I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Forcing a CODEC
Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing a CODEC
That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing a CODEC
Remove all other codec On Nov 15, 2011 8:17 PM, Jaap Winius jwin...@umrk.nl wrote: Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing a CODEC
The variable for outbound is (SIP_CODEC_OUTBOUND=g722) But I think asterisk will try to transcode then because the preferred codec on the phone is ulaw or so -Original Message- From: Danny Nicholas da...@debsinc.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 15 Nov 2011 08:50:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Forcing a CODEC That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Frequent Asterisk Restarts
On Tue, 2011-11-15 at 16:38 +0200, Tzafrir Cohen wrote: On Thu, Nov 10, 2011 at 03:02:43PM -0500, eherr wrote: Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my backtrace.txt completely useless or should I still submit? Don't bother. It makes the issue more aparent, but has a very large performance hit. In some cases it will also make the problem go away (in some odd races). Sorry, just need to double check I understand what you're saying here. Are you saying that compiling with the DONT_OPTIMIZE (and BETTER_BACKTRACE) flags causes a big performance hit to the asterisk service? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More than one route to a destination
Hi, I have a setup with 5 remote offices, each having a Asterisk PBX. I then have a central office, also with an Asterisk PBX. The remote offices have 2 links to the central office, a large link, and a smaller, but more reliable link. Unfortunately, using IAX is not an option for me. Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. Can you do this priority based on the user ID of the caller? Another question: If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? Kind Regards James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
can I make call without registration to an registered SIP account? -- Yes, you can but first you need to set allowguest=yes in sip.conf (makes ur server insecure) I guess you can put in same user/sip account in all iphones and like (in x-lite) don't let the phones register to server rather set the server IP as outbound proxy. /Sammy On Tue, Nov 15, 2011 at 7:40 PM, Faraj Khasib fkha...@iconnecths.comwrote: btw the call is one direction from clients to Call center My question can be rephrased can I make call without registration to an registered SIP account? From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [ fkha...@iconnecths.com] Sent: Tuesday, November 15, 2011 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple SIP endpoint registrations I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [ kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one route to a destination
IMO you can do this (I have a 1.4 client with 3 SIP trunks). Call-limit (or whatever flavor of that is applicable to your version) will let you control the flow across the trunks. The priority dialing would most likely have to be accomplished via AGI dialing since you would have to know if (a) a line is open or (b) can I kill the receptionists call so the boss can use the line? As for the last question, that's just some dialplan manipulation. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Courtier-Dutton Sent: Tuesday, November 15, 2011 9:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] More than one route to a destination Hi, I have a setup with 5 remote offices, each having a Asterisk PBX. I then have a central office, also with an Asterisk PBX. The remote offices have 2 links to the central office, a large link, and a smaller, but more reliable link. Unfortunately, using IAX is not an option for me. Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. Can you do this priority based on the user ID of the caller? Another question: If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? Kind Regards James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one route to a destination
Hi, Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? Yes. I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. 1- find out the criteria for Imp calls and write dialplan to use the reliable link and use other SIP trunk otherwise. Can you do this priority based on the user ID of the caller? Yes. For any outbound call see who is the caller and if CALLERID(num) matches use desired link. If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? AFAIK, you need to use DUNDI between the Asterisk Servers on top of SIP trunks. Once DUNDI is setup your users can move between offices and have just one extension. Regards, Sammy On Tue, Nov 15, 2011 at 8:12 PM, James Courtier-Dutton james.dut...@gmail.com wrote: Hi, I have a setup with 5 remote offices, each having a Asterisk PBX. I then have a central office, also with an Asterisk PBX. The remote offices have 2 links to the central office, a large link, and a smaller, but more reliable link. Unfortunately, using IAX is not an option for me. Can I use 2 SIP Trunks from each remote offices to the central site and permit 2 simultaneous calls across the SIP trunk that passes over the smaller line, and permit 10 simultaneous calls across the larger link? I also wish to have priorities, so that more important calls are sent over the smaller link (but more reliable) and the larger link used for less important calls. Can you do this priority based on the user ID of the caller? Another question: If a user with a SIP client starts off in remote office1, and then moves to remote office4, can then keep the same phone number? Kind Regards James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 SIP_CAUSE performance regression
Hi, We're using it here. As Ido asked, is there an alternative way of getting the SIP response in the event a Dial() fails? Cheers, Kingsley. On Thu, 2011-08-18 at 07:42 -0500, Matthew Nicholson wrote: Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,chan name)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Alternatively, with this feature enabled, anyone upgrading from Asterisk 1.4 will see a 60% decrease in the amount of SIP traffic they can handle before encountering problems. Before disabling this feature, we wanted to get a feel for how many people are using it. If you use this feature, please respond to this email and let us know. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Standard UIDs, especially for asterisk?
I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On 11/15/2011 09:58 AM, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? Cheers Tony There are no standard UID/GIDs for things. They are just system users that have no login shell. They are given lower IDs than normal user accounts (on redhat systems, see -r option to useradd) so that they can be easily distinguished. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On Tue, 15 Nov 2011, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. Different distros and different sysadmins have their own ideas about what numbers to use - I used to use 80 for the apache web user, but Debian for some weird reason likes 33 for example... 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? No. You may find that CentOS has an idea of what UIDs it likes to reserve for 'system' processes vs. users... See the man page for useradd (-r option) or adduser (--system option) depending on which one you prefer. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
In article 4ec28e0b.20...@digium.com, Jason Parker jpar...@digium.com wrote: On 11/15/2011 09:58 AM, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? Cheers Tony There are no standard UID/GIDs for things. They are just system users that have no login shell. They are given lower IDs than normal user accounts (on redhat systems, see -r option to useradd) so that they can be easily distinguished. Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. But it sounds like it is distro-specific. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On Tue, Nov 15, 2011 at 04:42:05PM +, Tony Mountifield wrote: But it sounds like it is distro-specific. No, it's system-specific. Debian for example will assign UIDs out of the relevant range based on the order in which packages are installed. Just use the textual UID/GID values, not the numeric ones. Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
In article alpine.deb.2.00.151609440.26...@unicorn.drogon.net, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 15 Nov 2011, Tony Mountifield wrote: I see on my CentOS systems that certain users for particular subsystems have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74. My two questions are: 1. Is there a list of these standard assignments somewhere? Googling did not turn up anything for me. Different distros and different sysadmins have their own ideas about what numbers to use - I used to use 80 for the apache web user, but Debian for some weird reason likes 33 for example... Ah, interesting. CentOS uses 48 for apache, so it evidently does vary between distros. 2. Are there standard values of UID and GID reserved for the asterisk user, if used for running Asterisk as non-root.? No. You may find that CentOS has an idea of what UIDs it likes to reserve for 'system' processes vs. users... See the man page for useradd (-r option) or adduser (--system option) depending on which one you prefer. Yes, I had been hoping there was a system UID reserved for asterisk, but apparently not. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
In article 4ec296b9.8040...@digium.com, Jason Parker jpar...@digium.com wrote: On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. If that were the case, I would expect different installations of the same distro (with varying package selections) to have different values for UIDs of specific system users. But examination of several different RH-based systems from FC1 through to CentOS 6 shows the same values being used. I would be reluctant to label all such packages as improperly written :-) Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Standard UIDs, especially for asterisk?
On Tue, 15 Nov 2011, Tony Mountifield wrote: In article 4ec296b9.8040...@digium.com, Jason Parker jpar...@digium.com wrote: On 11/15/2011 10:42 AM, Tony Mountifield wrote: Yes, I was hoping to use such a system user and group for asterisk, which would not conflict with any other system package I might install in the future, by virtue of being reserved for asterisk. There shouldn't be any conflict either way. (Properly written) packages don't specify a UID to use - they just get created sequentially, so the next available ID is used. If that were the case, I would expect different installations of the same distro (with varying package selections) to have different values for UIDs of specific system users. But examination of several different RH-based systems from FC1 through to CentOS 6 shows the same values being used. I would be reluctant to label all such packages as improperly written :-) I suspect the distros (well, Debian at least) have a standard 'skeleton' password file which they consider the minimum usable as part of the basic system, then packages added after the basic installation just get the next free number. Debian seems to install (for example) www-data (33 for as long as I can remember), games, man, lp, gnats, Debian-exim and a few others even when not using them at all! So if you're building a distro, then create one yourself, or if a package then use whatever the underying OS uses to pick the next system one. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto Queue, does not work, it should play message or any thing
On Tue, Nov 15, 2011 at 4:56 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten = 5631040,1,Playback(WelcomeMessage) exten = 5631040,2,Goto(OrangeCMG,s,1) So how I can overcome this? Show us the CLI output of a call that's not doing what you want and a call that is, and we can compare the differences. My guess is it has something to do with Playback having an automatic Answer(), and whatever you're Goto'ing doesn't... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto Queue, does not work, it should play message or any thing
Because playback is forcing an answer() before it starts; goto does not (no implied media need). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Tuesday, November 15, 2011 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Goto Queue, does not work, it should play message or any thing On Tue, Nov 15, 2011 at 4:56 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; When the call coming via the E1 dahdi and I handle the call (as first step) by exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be disconnected instead of queued. But, when I handle the call (as first step) by playing any sound file and then send for the queue, then it is working fine, WHY? exten = 5631040,1,Playback(WelcomeMessage) exten = 5631040,2,Goto(OrangeCMG,s,1) So how I can overcome this? Show us the CLI output of a call that's not doing what you want and a call that is, and we can compare the differences. My guess is it has something to do with Playback having an automatic Answer(), and whatever you're Goto'ing doesn't... -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Thanks for the answer Danny. Can you give an example of when we would setup peers through Method 1 2 as you described? If I am using FreePBX setup generic SIP extensions then use Polycom phones configure them to register with the SIP server (asterisk) with the extension/user password, are these extensions (or maybe more correctly peers) I setup via FreePBX actually self-registration or required-registration peers? Thanks, - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 From: Danny Nicholas [mailto:da...@debsinc.com] Sent: Monday, November 14, 2011 3:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Just trying to offer a little enlightenment - There are basically two methods of sip phone (peer/extension) registration. Method 1 is self-registration where Asterisk does not know or care about the phone until it asks to register. Method 2 is required-registration where Asterisk expects the phone to be there pretty much 24/7 and will attempt to register the phone and verify that it is still there at whatever frequency is specified. I personally record method 1 phones in users.conf and method 2 phones in sip.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Monday, November 14, 2011 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not mistaken. --E From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 14, 2011 5:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.commailto:d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]mailto:[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.comhttp://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
OK. Thanks everyone for the responses. If I can summarize, I think here's what's been discussed: Asterisk becomes aware of SIP extensions/peers, as soon as they register. Regarding how asterisk becomes aware of (or determines) that they are unavailable/unreachable, I believe I am hearing two possible scenarios: 1. The Interval of Registration. So asterisk has a timeout value that it is expecting the phone to reregister within. If the phone does not reregister within the timeout period, then asterisk determines that the extension/peer is no longer available. A few questions I have on this are: a. Where does this timeout interval come from? Is it a configuration parameter that we configure asterisk with, or is it something that is dynamically determined, or is it something that the phone/peer actually dictates to asterisk? b. If it is an asterisk configuration parameter, where does it exist (how do I set it confirm what it is currently set to)? It is a per-extension/peer setting, or is it global? c. Is there a command I can issue from the asterisk CLI to query it? 2. qualify=yes can be configured for any given SIP peer in asterisk. This will send a SIP OPTIONS message/packet to the peer every 1 or 2 minutes (depending on the configuration) that probes the peer to confirm it is still online. The keepalives (SIP OPTIONS packets) are actually sent from asterisk to the SIP peer, correct? But then the SIP peer actually has to respond to each one with its own SIP packet, correct? With this scenario, asterisk will still utilize scenario 1 (reregistration) as a means of determining that the peer is available, but additionally will continue to monitor the peer constantly (every 1-2 seconds) via these keepalives? This way asterisk is able to have a much more rapid discovery of peers that become unavailable (because they are literally no longer reachable, as they're no longer responding to the keepalives), correct? So my next questions are: a. Am I wrong with any of the above interpretations of the explanations you guys have given? b. Is the no-reply timer Sammy mentioned [(max time)x(max retries)] a parameter that can be set within asterisk? If so, what are the corresponding configuration parameters called? If not, what are the max time and max retries values? c. Is the SIP response the peer is supposed to give also an OPTIONS packet or something else? Thanks a LOT! I really appreciate all of the input insight you guys bring! - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 From: Sammy Govind [mailto:govoi...@gmail.com] Sent: Monday, November 14, 2011 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do extensions stay registered Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost) so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE. qualify=no wouldn't do all of the above. Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. On Tue, Nov 15, 2011 at 3:35 AM, eherr email.eherr9...@gmail.commailto:email.eherr9...@gmail.com wrote: I think the wrap up answer is the interval of registration compacted, if used, with the SIP OPTION packet. I like the SIP OPTION packet because we have scripts to monitor the status and lets us know when a phone is up or down. --E From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Monday, November 14, 2011 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do extensions stay registered I think the registration part was answered. The de-registration part is different. If the phone is gracefully taken off line it specifically de-registers. If it just can't be reached because it powers off or the router closes NAT, or whatever, then Asterisk won't know this until it times out. On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.commailto:email.eherr9...@gmail.com wrote: I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not
[asterisk-users] Asterisk Send out SIP Invites to external network- howto
Hello, Is there a way Asterisk can be used to send out SIP invites to external Network Gateways? I.E., I have an Asterisk with some softphones registered on it. I simply want to send out SIP invite, as simple as sip:call...@domain.com;transport=tcp, to an external Gateway, which will in turn establish a call by accepting it. I dont want my external network gateways to register with my Asterisk. In other words, I am asking if Asterisk can be asked to behave as a Endpoint? -- Thank you... Amar Akshat Please excuse any spelling mistakes, as this email was sent from a not so good mobile device. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forcing a CODEC
Hi Thats is also one of the reason On Tue, Nov 15, 2011 at 20:27, isr...@gmail.com wrote: The variable for outbound is (SIP_CODEC_OUTBOUND=g722) But I think asterisk will try to transcode then because the preferred codec on the phone is ulaw or so -Original Message- From: Danny Nicholas da...@debsinc.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 15 Nov 2011 08:50:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Forcing a CODEC That's one of the uses of the SIP_CODEC dialplan variable. Just set it in the context or the sip.conf or users.conf. In your particular case, just set up a specific context for the IAX calls [iax-in] Exten = _X.,1,Set(SIP_CODEC=G722) Exten = _X.,n,answer() -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius Sent: Tuesday, November 15, 2011 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Forcing a CODEC Hi folks, How can I take advantage of a high-bandwidth CODEC, like G.722, for internal communications at my site, but use G.711 (alaw/ulaw) for all other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX trunks, I was hoping that it would be possible to specify alaw and ulaw as the first two CODEC choices for the SIP phones, as well as in their sip.conf configurations, but that I could use the IAX trunks (with bandwidth=high) to force the phones to use their third CODEC choice, g722, because that would be the only CODEC specified for the IAX trunks (following disallow=all). Unfortunately, that doesn't work. Although the Asterisk console reports that g722 is being used, when I listen to the connection it's obvious that a G.711 CODEC is being used. Curiously, the reverse does work: if g722 is specified as the first CODEC of choice for the phones, it is possible to use the IAX trunks to force them to use alaw/ulaw instead. Is a solution to this problem? I'm using Debian squeeze with Asterisk 1.6.2.9. Cheers, Jaap -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Send out SIP Invites to external network- howto
Hi This can be done. On Wed, Nov 16, 2011 at 10:36, Amar Akshat amar.aks...@gmail.com wrote: Hello, Is there a way Asterisk can be used to send out SIP invites to external Network Gateways? I.E., I have an Asterisk with some softphones registered on it. I simply want to send out SIP invite, as simple as sip:call...@domain.com;transport=tcp, to an external Gateway, which will in turn establish a call by accepting it. I dont want my external network gateways to register with my Asterisk. In other words, I am asking if Asterisk can be asked to behave as a Endpoint? -- Thank you... Amar Akshat Please excuse any spelling mistakes, as this email was sent from a not so good mobile device. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Hey, I haven't thoroughly read the whole of your reply- just a quick answer to your timers question-generally I think you're right. Those timers are property of UAC so you may need to look into the phone configurations. I'd CISCO 79X0 phones and we wanted those to refresh their registrations at very short intervals of time as well as the INVITES timers was reduced too,...umm..I think that was for DNS-SRV based failovers. Though reducing the default timers from UAC heavily increased SIP traffic but we achieved the target by reducing the SIP timers in all phones. So that was an example. When you are using Asterisk as UAC to register onto another SIP server you can change the registration timeout and retry variables..and yes you can change these SIP timers in Asterisk sip.conf but thats not recommended.(see sip.conf.sample for details too) PS: with a quick look at sip.conf.sample + voip-info.org sip.conf details + google you can find lot more information than what you've collected so far. -- BR, Sammy On Wed, Nov 16, 2011 at 6:11 AM, Douglas Mortensen d...@impalanetworks.comwrote: OK. Thanks everyone for the responses. If I can summarize, I think here’s what’s been discussed: ** ** Asterisk becomes aware of SIP extensions/peers, as soon as they register.* *** ** ** Regarding how asterisk becomes aware of (or determines) that they are unavailable/unreachable, I believe I am hearing two possible scenarios:*** * ** ** **1. **“The Interval of Registration”. So asterisk has a timeout value that it is expecting the phone to reregister within. If the phone does not reregister within the timeout period, then asterisk determines that the extension/peer is no longer available. A few questions I have on this are: **a. **Where does this “timeout” interval come from? Is it a configuration parameter that we configure asterisk with, or is it something that is dynamically determined, or is it something that the phone/peer actually dictates to asterisk? **b. **If it is an asterisk configuration parameter, where does it exist (how do I set it confirm what it is currently set to)? It is a per-extension/peer setting, or is it global? **c. **Is there a command I can issue from the asterisk CLI to query it? **2. **“qualify=yes” can be configured for any given SIP peer in asterisk. This will send a SIP OPTIONS message/packet to the peer every 1 or 2 minutes (depending on the configuration) that probes the peer to confirm it is still online. The keepalives (SIP OPTIONS packets) are actually sent from asterisk to the SIP peer, correct? But then the SIP peer actually has to respond to each one with its own SIP packet, correct? With this scenario, asterisk will still utilize scenario 1 (reregistration) as a means of determining that the peer is available, but additionally will continue to monitor the peer constantly (every 1-2 seconds) via these keepalives? This way asterisk is able to have a much more rapid discovery of peers that become unavailable (because they are literally no longer reachable, as they’re no longer responding to the keepalives), correct? So my next questions are: **a. **Am I wrong with any of the above interpretations of the explanations you guys have given? **b. **Is the “no-reply” timer Sammy mentioned [(max time)x(max retries)] a parameter that can be set within asterisk? If so, what are the corresponding configuration parameters called? If not, what are the “max time” and “max retries” values? **c. **Is the SIP response the peer is supposed to give also an OPTIONS packet or something else? ** ** Thanks a LOT! I really appreciate all of the input insight you guys bring! ** ** - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 ** ** *From:* Sammy Govind [mailto:govoi...@gmail.com] *Sent:* Monday, November 14, 2011 10:36 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How do extensions stay registered ** ** Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost) ** ** so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE.* *** ** ** qualify=no wouldn't do all of the above. ** ** Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. ** ** On Tue, Nov 15, 2011
