Francisco Perez-Landaeta
2004-Jul-19 18:26 UTC
[Asterisk-Users] Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
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@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Mon, 19 Jul 2004 19:48:02 -0500 > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users digest, Vol 1 #4610 - 12 msgs > > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > 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@jovianprojects.com.au> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring. > Reply-To: asterisk-users@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@jovianprojects.com.au > (Move to Jupiter) > > --__--__-- > > Message: 2 > From: "Jason Kawakami" <jkkawakami@optellabs.com> > To: <asterisk-users@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@lists.digium.com > > Date: Mon, 19 Jul 2004 14:54:44 -0500 > From: "Christopher L. Wade" <clwade@sparco.com> > Organization: Unistar-Sparco Computers, Inc. > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration > Reply-To: asterisk-users@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@e2020inc.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@netgeeks.net]=20 > Sent: Monday, July 19, 2004 3:03 PM > To: asterisk-users@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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > From: Ryan Courtnage <ryan@voxbox.ca> > Organization: Coalescent Systems Inc > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] uip200 clips audio prompts > Date: Mon, 19 Jul 2004 17:28:47 +0000 > Reply-To: asterisk-users@lists.digium.com > > On July 19, 2004 10:19 pm, jparr@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@promedicalinc.com> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] MWI - Config Stupidity or Notify Issues? > Reply-To: asterisk-users@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@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@e2020inc.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@boltblue.com]=20 > Sent: Monday, July 19, 2004 2:11 PM > To: asterisk-users@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@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: 7 > Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > Date: Mon, 19 Jul 2004 16:41:58 -0700 > From: "Wiley E. Siler" <wsiler@e2020inc.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@sscsinc.com]=20 > Sent: Monday, July 19, 2004 3:35 PM > To: asterisk-users@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@mindworks.net]=20 >> Sent: Monday, July 19, 2004 10:26 AM >> To: asterisk-users@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@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> =20 >> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 8 > Date: Mon, 19 Jul 2004 17:07:02 -0700 > To: asterisk-users@lists.digium.com > From: "Chris A. Icide" <chris@netgeeks.net> > Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > Reply-To: asterisk-users@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@swtravel.com> > To: <asterisk-users@lists.digium.com> > Date: Mon, 19 Jul 2004 17:12:53 -0700 > Subject: [Asterisk-Users] Echo on a PRI > Reply-To: asterisk-users@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@radiumtec.com> > To: asterisk-users@lists.digium.com > Date: Mon, 19 Jul 2004 20:57:42 -0400 > Subject: [Asterisk-Users] Suscription > Reply-To: asterisk-users@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@e2020inc.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@swtravel.com]=20 > Sent: Monday, July 19, 2004 5:13 PM > To: asterisk-users@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@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@tootai.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] SIP to H323 call timeout > Reply-To: asterisk-users@lists.digium.com > > Fred Lee a ?crit : > >> >> >> 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@10.10.1.170;ftag=661806388;lr=on> >> Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: >> <sip:999012020@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@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@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@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@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@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@10.10.1.13' of Response 1: Found >> >> >> >> CONSOLE Output >> =============>> *CLI> -- Executing Dial("SIP/-08121388", >> "h323/15613021234@10.10.1.12:1720|60|C") in new stack >> -- Called 15613021234@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 >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >
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