[Asterisk-Users] Setup for Go2call ? Or any SIP provider using phonejack or
linejack g729 g723
Francisco Perez-Landaeta
fplandae at hotmail.com
Mon Jul 19 18:26:21 MST 2004
Hi, does anyone have the setup for go2call ?
I have digium boards and quicknet linejacks and phonejacks.
The cards work fine in asterisk without the g729 or g723.1 for the
phonejack.
I will like to do SIP origination using the codec in the phonejack and
linejack g729 or g723 and send the calls to go2call.
Anyone has the setup for this ? Or similar setup to a SIP provider using
g729 or g723
Thanks,
> From: asterisk-users-request at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
> Date: Mon, 19 Jul 2004 19:48:02 -0500
> To: asterisk-users at lists.digium.com
> Subject: Asterisk-Users digest, Vol 1 #4610 - 12 msgs
>
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> Today's Topics:
>
> 1. Re: Re: Cisco 7960 SIP V6 and distinctive ring. (Sam Tilders)
> 2. Re: Asterisk + NEC Electra Elite IPK Integration (Jason Kawakami)
> 3. RE: Polycom IP 500 Voicemail (Wiley E. Siler)
> 4. Re: uip200 clips audio prompts (Ryan Courtnage)
> 5. MWI - Config Stupidity or Notify Issues? (Robert Jackson)
> 6. RE: RE:RE: [Asterisk-Users] Codecs - Advantages (Wiley E. Siler)
> 7. RE: Polycom IP 500 Voicemail (Wiley E. Siler)
> 8. RE: Polycom IP 500 Voicemail (Chris A. Icide)
> 9. Echo on a PRI (David Goldfein)
> 10. Suscription (Carlos Clemares)
> 11. RE: Echo on a PRI (Wiley E. Siler)
> 12. Re: SIP to H323 call timeout (administrator tootai)
>
> --__--__--
>
> Message: 1
> Date: Tue, 20 Jul 2004 09:25:11 +1000
> From: Sam Tilders <sam at jovianprojects.com.au>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
> Reply-To: asterisk-users at lists.digium.com
>
> On Mon, Jul 19, 2004 at 02:09:34PM -0700, asteriskstuff @ ziplip. com wrote:
>> Thanks..it's a numeric value!! in the wiki it refers to a text field!!
>
> The wiki is also correct...
>
> I have:
> exten => 101,1,SetVar(ALERT_INFO=Bellcore-dr1)
>
> And that works fine.
>
> What was the error message you were getting?
>
> --
> --
> Sam Tilders
> sam at jovianprojects.com.au
> (Move to Jupiter)
>
> --__--__--
>
> Message: 2
> From: "Jason Kawakami" <jkkawakami at optellabs.com>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
> Date: Mon, 19 Jul 2004 17:28:09 -0600
> Reply-To: asterisk-users at lists.digium.com
>
> Date: Mon, 19 Jul 2004 14:54:44 -0500
> From: "Christopher L. Wade" <clwade at sparco.com>
> Organization: Unistar-Sparco Computers, Inc.
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
> Reply-To: asterisk-users at lists.digium.com
>
> Would the TLI(2)-U10 ETU work as well?
>
> That is a 2 port analog tie line card, I don't think that Digium has a card
> that can be set up as an analog 4W E&M trunk.
>
> bad idea anyway, the t-1 will be a much better interface and if you ever
> press the eject on the IPK you could use the t-1 as a PSTN interface.
>
>
> --__--__--
>
> Message: 3
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
> Date: Mon, 19 Jul 2004 16:28:25 -0700
> From: "Wiley E. Siler" <wsiler at e2020inc.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> Mine does the same. Once in Message center I can choose selection
> 1.Message Center and then soft key Select. Then I select the
> registered line that I want to check voice mail on. That is no less than
> 4 key strokes just to get into your voice mail. Not many to me but tons
> to an unskilled user. However, in the documentation regarding the
> bypassInstantMessage value, supposedly, setting bypassInstantMessage to
> 1 is supposed to allow you to go right into voice mail without
> navigating the Message Center. That is the big question on my mind at
> this point. I have yet to get this to work and I also don't think I am
> receiving any SIMPLE messages ti show me that I have messages waiting.
>
> Do you get a message waiting indicator?
>
> W
>
> -----Original Message-----
> From: Chris A. Icide [mailto:chris at netgeeks.net]=20
> Sent: Monday, July 19, 2004 3:03 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
>
> On 12:40 PM 7/19/2004, Wiley E. Siler wrote:
>> My Polycom is on loan as a demo and I assume it is one of the first
>> revision models. In fact it shows as Rev A on the back of the phone.
>>
>> I have all the same buttons you listed save for the Messages button.
>> The 3rd from the bottom on the right column of buttons sayd Voice Mail
>> on my version. That corresponds to the location of your button that
>> says Messages. I assume this was changed by Polycom since their phone
>> has other messaging capability (isntant message for instance) and it
> was >easier to use Messages and unify the meaning instead of Voice Mail
> and >lock it into one type of messaging.
>>
>> Does your Messages button dump you right into voice mail or do you
> have >to navigate a menu first?
>>
>> Thanks,
>> Wiley
>
> My messages button dumps me right to message center, which I then have
> to use soft buttons. My IP500 is Rev. C
>
>
> -Chris
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
> --__--__--
>
> Message: 4
> From: Ryan Courtnage <ryan at voxbox.ca>
> Organization: Coalescent Systems Inc
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] uip200 clips audio prompts
> Date: Mon, 19 Jul 2004 17:28:47 +0000
> Reply-To: asterisk-users at lists.digium.com
>
> On July 19, 2004 10:19 pm, jparr at bgcfreedom.com wrote:
>> On Mon, 19 Jul 2004, Ryan Courtnage wrote:
>>>> This happens with my 7940s as well. I have found that using and Answe=
> r,
>>>> and a Wait(1) before playing back prompts works well. Prevents Alisson
>>>> from saying "Assword?" when dialing VoicemailMail(20).
>>>
>>> Thanks for your reply. =A0I have been able to use this method to elimin=
> ate
>>> some of the problems, but from within the voicemail application, I don't
>>> beleive there is a way to set a delay between each prompt?
>>>
>>> ie: I'll hear:=A0"Press 0 for New messages, ... for old messages, ... f=
> or
>>> work message ....". =A0 The "Press x.." is cut off of the beginning of =
> the
>>> prompts.
>>>
>>> I only see this problem with uip200s. BT102s, handytones, sipuras, etc
>>> work just fine.
>>
>> Could it be silence supression?
>
> Perhaps. If the phone does support silence suppression, it isn't advertise=
> d -=20
> and neither are the config parameters needed to adjust it / turn it off.
>
> I'll check with Uniden.
> Thanks
> Ran
>
> --__--__--
>
> Message: 5
> Date: Mon, 19 Jul 2004 19:34:31 -0400
> From: "Robert Jackson" <RobertJ at promedicalinc.com>
> To: <asterisk-users at lists.digium.com>
> Subject: [Asterisk-Users] MWI - Config Stupidity or Notify Issues?
> Reply-To: asterisk-users at lists.digium.com
>
> I am having a problem with the message waiting indicator. We are
> currently using the ast_data modules for both our sip configuration and
> our voicemail configuration. In the mailbox field I have tried using
> both mailboxnumber at context and simply mailboxnumber. Yet so far I am
> still not getting a MWI on my 7905's or on my 7960's. My assumption
> would be that I am still missing something, but at this point I can't
> figure it out. I have recently seen a message that Notify is not
> working properly with CVS HEAD. =20
>
> Thanks for you help in advance.
>
> Robert Jackson
>
> --__--__--
>
> Message: 6
> Subject: RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
> Date: Mon, 19 Jul 2004 16:32:43 -0700
> From: "Wiley E. Siler" <wsiler at e2020inc.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> Is this bascially setting your bandwith value =3D high inside of =
> iax.conf?
>
> Or is there another place to designate the codec?
>
> Thanks,
> Wiley
> =20
>
> -----Original Message-----
> From: Senad Jordanovic [mailto:senad at boltblue.com]=20
> Sent: Monday, July 19, 2004 2:11 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs -
> Advantages
>
> Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30
> calls. I do have issues with processing CPU capacity. Is g711 CPU
> intensive as g729 ? I understand g729 is very CPU intensive.
>>>> .......
>
> Forgive me, but what you just wrote tells you EXACTLY what you should
> use!
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 7
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
> Date: Mon, 19 Jul 2004 16:41:58 -0700
> From: "Wiley E. Siler" <wsiler at e2020inc.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> Thank you so much! That was exactly what I needed to know!
>
> Cheersm
> Wiley
> =20
>
> -----Original Message-----
> From: Tor Roberts [mailto:voip at sscsinc.com]=20
> Sent: Monday, July 19, 2004 3:35 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
>
> Wiley,
> I don't have any 500s, but I use 600s, which use the same file I think.=20
> Here is my digitmap:
>
> <digitmap
> dialplan.digitmap=3D"9[2-9]xxxxxx|91xxxxxxxxxx|85xx|[5-7]xx|9411|9911"=20
> dialplan.digitmap.timeOut=3D"3"/>
>
> What this says is that if I dial 9, then a 7 digit local number, I
> don't need to hit send. If I dial 91, then 10 digit long distance
> number, I don't need to hit send. If I dial extension 85 plus any 2
> digits ex., 8523, I don't need to hit send. If I dial extension 5,6, or
> 7 plus 2 digits, ex. 635, I don't need to hit send. And if I dial 9411,
> or 9911 (info or emergency) I don't need to hit send.
> Hope this helps.
>
> -Tor
>
> Wiley E. Siler wrote:
>
>> I read the administrator document repeatedly. I have not been able to
>> find a wiki that applied to digitmap feature at all and I have searched
>> repeatedly and read several of the wikis regarding Polycoms. The
>> administrators guide doesn't have enough context explanation to make
> the
>> use of the digitmap understandable.=20
>>
>> That is the basis of my request for a digitmap explanation. I am not
>> asking someone to write mine for me. I am asking to see an example and
>> an explanation that gives context so I can write my own and know I have
>> done it properly. My PBX is Asterisk and the setup is about as generic
>> as generic can be. Polycoms over SIP to the PBX.
>>
>> If you know where the wiki is for digitmaps please send it. If you
> feel
>> inspired, a short explanation of the relevance and context of digitmaps
>> would be greatly appreciated. I know everyone has to take their own
>> time to answer these emails and I truly appreciate that. That is why I
>> do my research until I hit a wall, then I will ask here. I appreciate
>> whatever you can spare time for.
>>
>> Thanks!
>> Wiley
>>
>> =20
>>
>> -----Original Message-----
>> From: Brent Franks [mailto:mwless at mindworks.net]=20
>> Sent: Monday, July 19, 2004 10:26 AM
>> To: asterisk-users at lists.digium.com
>> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
>>
>> =20
>>
>>> Thank you!
>>>
>>> Can you tell me more about the dial plan feature? How do you setup
>>> =20
>>>
>> the
>> =20
>>
>>> correct digitmap?
>>>
>>> =20
>>>
>>
>> Check the Administrator's Document. You can find it on the Wiki, under
>> IP Phones.. Polycom. Did you try to look up the digitmap feature
> before
>> sending this post? If not, you should be able to understand it when
> you
>> read it, it's relatively straight forward.
>>
>> No one can setup a correct digitmap for you, as it will vary greatly on
>> how you have setup your PBX.
>>
>> - Brent
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> =20
>>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 8
> Date: Mon, 19 Jul 2004 17:07:02 -0700
> To: asterisk-users at lists.digium.com
> From: "Chris A. Icide" <chris at netgeeks.net>
> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
> Reply-To: asterisk-users at lists.digium.com
>
> On 04:28 PM 7/19/2004, Wiley E. Siler wrote:
>> Mine does the same. Once in Message center I can choose selection
>> 1.Message Center and then soft key Select. Then I select the
>> registered line that I want to check voice mail on. That is no less than
>> 4 key strokes just to get into your voice mail. Not many to me but tons
>> to an unskilled user. However, in the documentation regarding the
>> bypassInstantMessage value, supposedly, setting bypassInstantMessage to
>> 1 is supposed to allow you to go right into voice mail without
>> navigating the Message Center. That is the big question on my mind at
>> this point. I have yet to get this to work and I also don't think I am
>> receiving any SIMPLE messages ti show me that I have messages waiting.
>>
>> Do you get a message waiting indicator?
>>
>
> I do get MWI, there are a few things you need to set, and I forget what off
> the top of my head, soon as I can look and post it here.
>
> I haven't tried the bypassInstantMessage value, but I'll take a look and
> see if I can get it to work.
>
> -Chris
>
>
> --__--__--
>
> Message: 9
> From: "David Goldfein" <dave at swtravel.com>
> To: <asterisk-users at lists.digium.com>
> Date: Mon, 19 Jul 2004 17:12:53 -0700
> Subject: [Asterisk-Users] Echo on a PRI
> Reply-To: asterisk-users at lists.digium.com
>
> Hi,
> I recently set up the following in a production system (2.8 GHZ Xeon, 1 =
> Gig
> Memory, Dell 2650).
>
> Telco - PRI - Asterisk - T1 - PBX
>
> I am getting an occasional noticeable echo on some of the phone lines
> (random inbound and outbound). Everyone I ask keeps telling me that I =
> can't
> be having echo since I am on a PRI, which is a digital circuit. Ok, so =
> I
> can't be having echo, but I am! Does anyone have any ideas of what =
> might be
> causing the echo in this situation? =20
>
>
> Thanks,
> Dave
>
>
>
> --__--__--
>
> Message: 10
> From: Carlos Clemares <cclemares at radiumtec.com>
> To: asterisk-users at lists.digium.com
> Date: Mon, 19 Jul 2004 20:57:42 -0400
> Subject: [Asterisk-Users] Suscription
> Reply-To: asterisk-users at lists.digium.com
>
> Name: Carlos Clemares
>
>
> --__--__--
>
> Message: 11
> Subject: RE: [Asterisk-Users] Echo on a PRI
> Date: Mon, 19 Jul 2004 17:27:32 -0700
> From: "Wiley E. Siler" <wsiler at e2020inc.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> I think I saw a reference to a similar problem and it regarded IRQ
> issues on the machine in question. IF there was IRQ sharing, cagey
> things happened. But if the T1 card had a static IRQ, it resolved the
> issue. Does your T1 card have a dedicated IRQ? I am sure someone will
> be able to explain further and possibly give you some validation on your
> Mobo too?
>
> Thanks,
> Wiley
>
>
> -----Original Message-----
> From: David Goldfein [mailto:dave at swtravel.com]=20
> Sent: Monday, July 19, 2004 5:13 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Echo on a PRI
>
> Hi,
> I recently set up the following in a production system (2.8 GHZ Xeon, 1
> Gig Memory, Dell 2650).
>
> Telco - PRI - Asterisk - T1 - PBX
>
> I am getting an occasional noticeable echo on some of the phone lines
> (random inbound and outbound). Everyone I ask keeps telling me that I
> can't be having echo since I am on a PRI, which is a digital circuit.
> Ok, so I can't be having echo, but I am! Does anyone have any ideas of
> what might be causing the echo in this situation? =20
>
>
> Thanks,
> Dave
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 12
> Date: Tue, 20 Jul 2004 02:34:31 +0200
> From: administrator tootai <admin at tootai.net>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] SIP to H323 call timeout
> Reply-To: asterisk-users at lists.digium.com
>
> Fred Lee a écrit :
>
>>
>>
>> Hi all,
>>
>> I have the following setup:
>>
>> UAs ------------SER ---------- ASTERISK ----------GNUGK --------- GWs
>>
>> SER is configured to route call requests from UAs to Asterisk.
>> Asterisk is configured to receive the call on SIP channel and dial out
>> to GNUGK over H323 channel. The problem I'm facing is that asterisk
>> sends out the call request to GNUGK and times out immediately, so call
>> setup is never completed. On GNUGK the call request comes in followed
>> by a normal call drop.
>>
>> Any ideas on what could be the problem ??
>
> Do you use the h323 - Nufone? Is it a recent installation? If so, could
> be the problem that GW need FastStart and the * h323 don't send it.
>
>>
>> My asterisk configuration, debug and console output are as follow :
>>
>> SIP.CONF
>> ======
>> [general]
>> port = 5080
>> bindaddr = 10.10.1.170
>> context = to_GNUGK
>> disallow=all
>> allow=g729
>>
>>
>> H323.CONF
>> =======
>> [general]
>> port = 1720
>> allow = g729
>> gatekeeper = 64.80.103.12
>> allowgkrouted = yes
>> context = to_SER
>>
>> EXTENSIONS.CONF
>> ============
>> [general]
>> static = yes
>> writeprotect = yes
>>
>> [to_GNUGK]]
>> exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C)
>>
>> [to_SER]
>> exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60)
>>
>>
>>
>> DEBUG File
>> ==========
>> Jul 15 16:14:10 DEBUG[65541]: Check for res for
>> Jul 15 16:14:10 DEBUG[65541]: is not a local user
>> Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop:
>> <sip:15613021234 at 10.10.1.170;ftag=661806388;lr=on>
>> Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop:
>> <sip:999012020 at 10.10.1.13:5060;user=phone;transport=udp>
>> Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL)
>> Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256,
>> data=15613021234 at 10.10.1.12:1720.
>> Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username:
>> 15613021234
>> Jul 15 16:14:10 DEBUG[311316]: dest=15613021234 at 10.10.1.12:1720,
>> timeout=0.
>> Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess
>> Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL)
>> Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256,
>> data=t at 10.10.1.12:1720.
>> Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t
>> Jul 15 16:14:23 DEBUG[311316]: dest=t at 10.10.1.12:1720, timeout=0.
>> Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess
>> Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL)
>> Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256,
>> data=h at 10.10.1.12:1720.
>> Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h
>> Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter
>> Jul 15 16:14:31 DEBUG[311316]: is not a local user
>> Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on
>> '842589597 at 10.10.1.13' of Response 1: Found
>>
>>
>>
>> CONSOLE Output
>> ==============
>> *CLI> -- Executing Dial("SIP/-08121388",
>> "h323/15613021234 at 10.10.1.12:1720|60|C") in new stack
>> -- Called 15613021234 at 10.10.1.12:1720
>> == No one is available to answer at this time
>>
>> -- Timeout on SIP/-08121388
>> == CDR updated on SIP/-08121388
>>
>> _________________________________________________________________
>> MSN 8 with e-mail virus protection service: 2 months FREE*
>> http://join.msn.com/?page=features/virus
>>
>> _______________________________________________
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --__--__--
>
> _______________________________________________
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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