[asterisk-users] problems getting chan_alsa.so to run
Hi!
I am interisted to dial out from the console with chan_alsa. Can
somebody of you help me according this problem?!
I added user the asterisk to pulse and pulse-access, and it didn't
change anything. alsa applications are routed by default to pulse.
cat /etc/asound.conf
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
What might be the problem?! Here is the ouptut:
office*CLI module load chan_alsa.so
Unable to load module chan_alsa.so
Command 'module load chan_alsa.so ' failed.
== Parsing '/etc/asterisk/alsa.conf': == Found
[Oct 20 21:54:16] ERROR[17849]: chan_alsa.c:180 alsa_card_init:
snd_pcm_open failed: Connection refused
[Oct 20 21:54:16] ERROR[17849]: chan_alsa.c:276 soundcard_init: Problem
opening alsa capture device
== No sound card detected -- console channel will be unavailable
== Turn off ALSA support by adding 'noload=chan_alsa.so' in
/etc/asterisk/modules.conf
Tamer
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Re: [asterisk-users] USA Did required
I say only, that it is MONEYMAKING ISDN is the best thing ever. Using the internet completly through VoIP is wasting ressources and if the ISP connection falls for a time, the line communication fall either. I think if a state permits ordering isdn bri channels, then there is only a thought behind to make more money for each lines that is being ordered or to sell for high prices ISDN PRI channels, which is in my view nothing else as a unfair isolation business policy. Sorry, but SMB need to work efficient to grow with telephony without falling in high expenses. I spoke with a friend of mine in L.A. and in the US professional telephony services are much more expensive as in Germany. I still don't know why. Tamer Am 01.10.2011 18:59, schrieb Eric Wieling: In the USA ordering BRI service is discouraged by the telcos and is very uncommon. In Verizon NE CLECs are not even permitted to order BRIs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: Friday, September 30, 2011 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] USA Did required Go to ATT and ask for a BRI ISDN Line. Tell them that you want a system access with a number block. that would be typically 54448[0-9] where your extensions are from 0-9 I don't know what protocol the americans are using, as I know the americans made for sure their own thing which is known as the north american BRI protocoll. If they have replaced it with the euro isdn protocol, I don't know. Figure out in advance if the supplier, in this case Digium, Sangoma or Beronet support the north american protocol. in Germany we use the EUROISDN protocol. Here in Germany it is usual to have such lines, we don't use VoIP with DID. logicly it could be possible, but I never saw a provider. By the ISDN is more secure. If ISP connections fall out, you are through your telephone lines still reachable. Information: For BRI Isdn you might need a NT Unit where you would connect the isdn cable to your board. Tamer Am 01.10.2011 02:29, schrieb Tarek Sawah: Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- -- Date: Sat, 1 Oct 2011 00:51:59 +0530 From: amit.magn...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] USA Did required Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- 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 -- _ -- 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] USA Did required
Go to ATT and ask for a BRI ISDN Line. Tell them that you want a system access with a number block. that would be typically 54448[0-9] where your extensions are from 0-9 I don't know what protocol the americans are using, as I know the americans made for sure their own thing which is known as the north american BRI protocoll. If they have replaced it with the euro isdn protocol, I don't know. Figure out in advance if the supplier, in this case Digium, Sangoma or Beronet support the north american protocol. in Germany we use the EUROISDN protocol. Here in Germany it is usual to have such lines, we don't use VoIP with DID. logicly it could be possible, but I never saw a provider. By the ISDN is more secure. If ISP connections fall out, you are through your telephone lines still reachable. Information: For BRI Isdn you might need a NT Unit where you would connect the isdn cable to your board. Tamer Am 01.10.2011 02:29, schrieb Tarek Sawah: Google is your best friend when looking for this type of assistance my friend. try callcentric vonage packet8 for reliable retail DIDs. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 1 Oct 2011 00:51:59 +0530 From: amit.magn...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] USA Did required Hello members, I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices. The DID will only be required to receive inbound calls and no outbound calls. Let me know your best per month prices/cost for the above. Regards, Amit Mehta -- _ -- 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] Digium ISDN card
compare the prices between sangoma and digium pri boards! Sangoma's oards here in Germany are cheaper as the ones from digium. if you need detailed help, you can contact me, and I can workout for you something as well as helping you setting up your pbx! Tamer Am 23.09.2011 15:01, schrieb michael k: Hi All, I am new in asterisk. In my office we have purchased ISDN pri line with 30 channels. we have more than 60 soft phone nodes and the internal asterisk connectivity between extensions are working with soft phones. Can anybody tell me which pci or pci express digium card can be used to connect my asterisk server and the ISDN pri line with 30 channels ? Please assist me to do if possible Michael.k -- _ -- 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] mISDN Vs Dahdi
Be sure, if you make us of HFC Boards that you have the zapfhfc patches. There is some work for you to accomplish, like patching dahdi to make use with the cheap isdn boards. For office using ISDN Devices it's fairly enough. If you want to make use of a server, I advise you to take the digium or sangoma boards, because of the native support for asterisk. If you don't love patching and searching to get what in the internet, take Gentoo Linux. All included (don't use dahdi 1.5.x, still not working on gentoo). Tamer Am 21.09.2011 12:43, schrieb Gopal krishnan: Hi Tamer, Many thanks for your comments, really your comments are useful. And finally I think using dahdi instead of mISDN is better. On Wed, Sep 21, 2011 at 3:10 AM, Tamer Higazi th9...@googlemail.com mailto:th9...@googlemail.com wrote: Am 20.09.2011 19:47, schrieb Gopal krishnan: What is the difference between using mISDN for BRI and using Dahdi mISDN was at 1st done for ISDN Services and channel driver as I know. It supported like call routing (switch based, not your side on the pbx level). without mISDN? you can use DAHDI without ISDN, for other related telephony interfaces. Like analogue cards. and if there are other telephony hardware interfaces that has nothing common todo you can use it too. and genereal: DAHDI is the native digium hardware telephony interface. Beside mISDN you have the native support for an echo cancellor you could use. for example: oslec or the hpec (high performance echo cancellor) which you can't make use of it natively with mISDN. As long you have no hardware interface boards as PCI modules, and you connect only throug the network to your enddevices, there is no need to startup dahdi at all. end beside: dahdi is an extra service that starts up, mISDN is a channel driver you must activate in the modules.conf. Are all your question answered that far?! Regards Tamer -- _ -- 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] Dahdi and BRI NT PtmP
2.4.x does it with me, so I am sure 2.5.x do makes it either! Tamer Am 21.09.2011 16:13, schrieb Olivier: Hello, Is Dahdi 2.5.0 supposed to support BRI NT PtmP mode ? Regards -- _ -- 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] mISDN Vs Dahdi
Am 20.09.2011 19:47, schrieb Gopal krishnan: What is the difference between using mISDN for BRI and using Dahdi mISDN was at 1st done for ISDN Services and channel driver as I know. It supported like call routing (switch based, not your side on the pbx level). without mISDN? you can use DAHDI without ISDN, for other related telephony interfaces. Like analogue cards. and if there are other telephony hardware interfaces that has nothing common todo you can use it too. and genereal: DAHDI is the native digium hardware telephony interface. Beside mISDN you have the native support for an echo cancellor you could use. for example: oslec or the hpec (high performance echo cancellor) which you can't make use of it natively with mISDN. As long you have no hardware interface boards as PCI modules, and you connect only throug the network to your enddevices, there is no need to startup dahdi at all. end beside: dahdi is an extra service that starts up, mISDN is a channel driver you must activate in the modules.conf. Are all your question answered that far?! Regards Tamer -- _ -- 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] ISDN2 PCIe Card for Asterisk
what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- € inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- 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] Beggining asterisk
I advise you taking Gentoo Linux. There is a great asterisk repisetory. Also support patvhes for NON digium hardware. Resources: http://oreilly.com/openbook/ there you find on the right the free book Asterisk: The Future of telephony the 3rd edition is available, but that book covers every thing to run the asterisk PBX. Tamer Am 03.09.2011 14:00, schrieb Daniel Tryba: On Sat, Sep 03, 2011 at 01:41:39AM -0400, Esteban Cacavelos wrote: Hi all, i am beggining on asterisk and i would like to run my asterisk on Ubuntu server 10.04 + asterisk 1.8.6.0 + dahdi [snip] I was looking for documentation and i found the official book, but it doesn't explain anything about Dahdi. Replace zap(tel) with dahdi. But I suggest starting with the Ubuntu bundeled packages (asterisk/dahdi/libpri). -- _ -- 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 on Android?
Do you want to run the entire PBX on the Android client or are you just looking for a IAX programm to be installed for receiving calls?! I think this is what you ment. Here is the url: https://market.android.com/details?id=com.bw.iax.ui Am 02.09.2011 16:32, schrieb Gilles: On Fri, 2 Sep 2011 13:23:18 +0100, A J Stiles asterisk_l...@earthshod.co.uk wrote: TTBOMK it's been done; but without the necessary Zaptel / DAHDI drivers to interface with the phone line, it's rather less useful than it sounds. I'm looking for a way to an IVR in my smartphone to handle incoming calls, and right it depending on such and such option. Anyone has more information in turning a smartphone (Android and/or iPhone) into a basic IP PBX? Thank you. -- _ -- 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] dahdi with isdn nt_mode, phone no signal still.
Hi people! I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on asterisk. On which DAHDI tells me also properly that both of my boards are registered, one in TE and the other on in NT mode. Calls do successfully come inside, but I want to connect my ISDN phone at the board, but the phone is death. What did I make wrong?! dahdi_scan: office tamer # dahdi_scan [1] active=yes alarms=OK description=HFC-S PCI A ISDN card 0 [TE] name=ZTHFC1 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 01 Slot 01 basechan=1 totchans=3 irq=17 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding=AMI framing=CCS [2] active=yes alarms=OK description=HFC-S PCI A ISDN card 1 [NT] name=ZTHFC2 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 01 Slot 02 basechan=4 totchans=3 irq=21 type=digital-NT syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding= framing=CAS office tamer # and dahd status in asterisk: office*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO HFC-S PCI A ISDN card 0 [TE] OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) HFC-S PCI A ISDN card 1 [NT] OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) TE-mode works fine so far. -- _ -- 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] dahdi with isdn nt_mode, phone no signal still.
Hi Patrick! Now i got it. I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1. The patches are automatically integrated at Gentoo. I didn't have to patch anything. That did the community. Another question, I really don't like to buy a new ISDN phone with external power connector, can I make the power supply for the phone somehow?! Tamer Am 30.08.2011 20:05, schrieb Patrick Lists: On 08/30/2011 06:32 PM, Tamer Higazi wrote: Hi people! I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on asterisk. On which DAHDI tells me also properly that both of my boards are registered, one in TE and the other on in NT mode. Calls do successfully come inside, but I want to connect my ISDN phone at the board, but the phone is death. What did I make wrong?! Afaik ISDN phones need power to work. You would need a HFC-S card with a power plug where you can hook up power from the computer's power supply so the card can power the phone. Like Sangoma's B700. Alternatively perhaps you could try to power the phone through an NT1? Would you mind sharing which version of DAHDI you used and where you got the HFC-S patch? Regards, Patrick -- _ -- 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] How Cisco ATA 186 through SCCP with skinny.conf ?!
Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer -- _ -- 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] voicemail problem
Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message please record your message after the tone and hang up or press the pound key. Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- _ -- 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] h323 guide for asterisk
Hi people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening 2 and more channels on 1 SIP account
D Tucny schrieb: 2009/4/18 Tamer Higazi th9...@googlemail.com mailto:th9...@googlemail.com Scenario: I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP cpu to take out the echo cancellation. Communication is done through the chan_capi interface module. If a call comes inside, and I forward it to the SIP account that is registered in the module, then all DECT phone do ring. But DECT / GAP phones are not designed for these issues. Scenario what a commercial PBX system does which has a ISDN board. Set up the phones: 1 - queues through system messages the dect man station on which the cordless devices are registered to. the main station tells him the ID of the devices and I assign through the webinterface the numbers (DDI or MSN) to the devices. 2 - set up is done! Call routine: Call in! 1 - from the NT unit of my home line comes a call that goes to the PBX. 2 - The PBX which receives the call extract the number (DDI or MSN) and compare it in the list of which phone it is (from step one)! 3 - The PBX send a message queue to the base station to check if the phone is busy, if yes forget it. If no pass the call through. Done with sending a message to the base that the call is passed to this device, for that the other devices won't ring. Call out! 1 - from the handset I make a call 2- the PBX, sends a message to the base station asking who dialed the number. 3 - the base station gives back the id, the outgoing number is set for that the call is passed through with the desired outgoing number. Now Asterisk, if SIP supports it receiving and placing several calls through one FXS port: the agi script: http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText 1 - a call is placed 2 - the agi script sends a message through the sip channel and the anser comes back, the answer is held in a variable 3 - the variable had been worked out, and the MSN or DDI is set --- 1 a call is received through the chan_capi interface 2. the dialplan knows which id belongs to the DDI or MSN number and calls the AGI script, which sends the message to the base station asking if the handset is available, busy or ready to receive calls. 3. the script returns a value that is being worked out and the agi script is called again to tell the base station that the incoming call is for the handset id (let us say number 5), that not all phones do ring. 4. the call is forwarded to the FXS port and that's it. This is how a usual PBX System in Germany and across europe do work. But if SIP or Asterisk do not support receiving and placing more calls through one FXS port and channel at the same time, then the DECT sollution can be dropped at all for me, and I shouldn't lose more time in this issue. DECT itself, is a well worked out technologie that gives you the chance to make a lot! It is programming work, not more then that. I hope all questions are being answered. You are confused... no I am not While DECT may well be capable of this sort of functionality and while asterisk, and SIP are capable of this sort of functionality, you are using an intermediate technology, a single POTS analogue connection, that isn't capable... read the DECT specification from A-Z. In Germany we have digital (isdn analog adapter that do it). By the way, in Germany and many other European countries, BRI ISDN connections for household and companies are widespread. commercial PBX systems are BRI ISDN and analogue FXS related. Call comes in, and call is going out. If you read the DECT specification completly! you know how to set up before you route the call the DECT / GAP system. You'll need a DECT base that either directly supports SIP for communicating with Asterisk, or, with a more capable interface, such as ISDN, that allows for more advanced communication and multiple channels... not true, read the specification from A-Z, i't only about SIP, to receive several calls at the same time, as well placing. The best you could probably hope to get using an FXS connection is that a single inbound call could be routed to one of the handsets by using distinctive ring, if the base supports it... However, you can not have more than one call over one analogue FXS connection, this isn't an Asterisk or SIP limitation, this is a limitation of the analogue connection... Example SIP DECT devices: http://www.snom.com/en/products/snom-m3-voip-phone/ http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-1814D885/03/hs.xsl/30395.htm Multiple Siemens devices d Tamer ___ -- Bandwidth
[asterisk-users] opening 2 and more channels on 1 SIP account
Hi! I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: 1. The Base station of the DECT cordless phones, is connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want to place and receive as many calls at the same time through 1 SIP Account, and through this 1 FXS Port where the base station is connected through. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening 2 and more channels on 1 SIP account
Danny Nicholas schrieb: This is an HT-486 (HT-488, HT-496)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, April 17, 2009 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] opening 2 and more channels on 1 SIP account On Fri, 17 Apr 2009, Tamer Higazi wrote: Sorry I write this message into the developer lailing list, I do this because nobody in the user list could answer me this question, due it's to technical. Date: Fri, 17 Apr 2009 18:51:17 +0200 To: asterisk-users@lists.digium.com Date: Fri, 17 Apr 2009 19:34:46 +0200 To: asterisk-...@lists.digium.com Impatient are we :) Yes you are entirely right! :) The dev list is for discussion that involves changing the C source code of Asterisk. I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: 1. The Base station of the DECT cordless phones, is connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want to place and receive as many calls at the same time through 1 SIP Account, and through this 1 FXS Port where the base station is connected through. Your assumption is incorrect. The question is not technical at all (assuming I understand correctly). Connecting a cordless base that has 2 handsets to an ATA does not magically give you 2 separate lines unless: ) The DECT base is a 2 line base. If so, does it have 2 rj-11's or 1? Since the other handset says busy this implies it is a single line DECT base. it has only 1 rj-11 port. Not true, with the DECT specification of ETSI I can send System messages where I can route the call from the base station to the handset direclty. DECT: http://www.etsi.org/WebSite/Technologies/DECT.aspx Standards EN-300 175 parts 1-8 (part 5) with system messages you can receive device through outgoing calling number and assign through ISDN a DDI or MSN number, like those commercial PBX. The messages can be send through a small 2 way AGI script. Now, if you guys tell me that SIP isn't able to make it, to receive more then one calls at the same time, as well as placing, even if the base station has only 1 RJ11 port, then cordless phone systems aren't suitable at all for the asterisk PBX, which is a very sad issue. But answering your question with a single DECT base system, that it is only capable to receive one call is NOT RIGHT. Commercial PBX vendors from example Auerswald, and clients (like the FritzBox 7130 from AVM) are doing making these tasks possible with the system messaging system of DECT defined at the European Telecomunication Standards Institute. I hope still, that SIP can manage more then one call through one account, if not, it is a sad thing and I have to drop the development for DECT for the PBX system at all. ) Your ATA supports 2 lines. If so, does it have 2 rj-11's or 1? You would need to provide a bit more information, but there is nothing in the information you have provided to indicate that this has anything to do with Asterisk. Maybe you would have more success calling the Grandstream or the manufacturer of your DECT set. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] opening 2 and more channels on 1 SIP account
Scenario: I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP cpu to take out the echo cancellation. Communication is done through the chan_capi interface module. If a call comes inside, and I forward it to the SIP account that is registered in the module, then all DECT phone do ring. But DECT / GAP phones are not designed for these issues. Scenario what a commercial PBX system does which has a ISDN board. Set up the phones: 1 - queues through system messages the dect man station on which the cordless devices are registered to. the main station tells him the ID of the devices and I assign through the webinterface the numbers (DDI or MSN) to the devices. 2 - set up is done! Call routine: Call in! 1 - from the NT unit of my home line comes a call that goes to the PBX. 2 - The PBX which receives the call extract the number (DDI or MSN) and compare it in the list of which phone it is (from step one)! 3 - The PBX send a message queue to the base station to check if the phone is busy, if yes forget it. If no pass the call through. Done with sending a message to the base that the call is passed to this device, for that the other devices won't ring. Call out! 1 - from the handset I make a call 2- the PBX, sends a message to the base station asking who dialed the number. 3 - the base station gives back the id, the outgoing number is set for that the call is passed through with the desired outgoing number. Now Asterisk, if SIP supports it receiving and placing several calls through one FXS port: the agi script: http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText 1 - a call is placed 2 - the agi script sends a message through the sip channel and the anser comes back, the answer is held in a variable 3 - the variable had been worked out, and the MSN or DDI is set --- 1 a call is received through the chan_capi interface 2. the dialplan knows which id belongs to the DDI or MSN number and calls the AGI script, which sends the message to the base station asking if the handset is available, busy or ready to receive calls. 3. the script returns a value that is being worked out and the agi script is called again to tell the base station that the incoming call is for the handset id (let us say number 5), that not all phones do ring. 4. the call is forwarded to the FXS port and that's it. This is how a usual PBX System in Germany and across europe do work. But if SIP or Asterisk do not support receiving and placing more calls through one FXS port and channel at the same time, then the DECT sollution can be dropped at all for me, and I shouldn't lose more time in this issue. DECT itself, is a well worked out technologie that gives you the chance to make a lot! It is programming work, not more then that. I hope all questions are being answered. Tamer Steve Edwards schrieb: On Sat, 18 Apr 2009, Tamer Higazi wrote: On Fri, 17 Apr 2009, Steve Edwards wrote: Impatient are we :) Yes you are entirely right! :) Patience is a virtue. If so, does it have 2 rj-11's or 1? Since the other handset says busy this implies it is a single line DECT base. it has only 1 rj-11 port. ... with the DECT specification of ETSI I can send System messages where I can route the call from the base station to the handset direclty. Isn't DECT (Digital Enhanced Cordless Telecommunications) the protocol between the handset and the base? From the base on, isn't it just POTS? with system messages you can receive device through outgoing calling number and assign through ISDN a DDI or MSN number, like those commercial PBX. The messages can be send through a small 2 way AGI script. The base has an rj-11 but is an ISDN device? I thought ISDN was rj-4[58]. If your base is ISDN shouldn't you be connecting it to an ISDN terminal adapter instead of an ATA? What the heck is a 2 way AGI script? To whom and from whom? Now, if you guys tell me that SIP isn't able to make it, to receive more then one calls at the same time, as well as placing, even if the base station has only 1 RJ11 port, then cordless phone systems aren't suitable at all for the asterisk PBX, which is a very sad issue. I think I'm confused. If the base has a single 2-wire rj-11, it's a single line POTS device regardless of how many handsets you have or the capabilities of your ATA. The fact that you say the not in use handset says busy supports my assumptions. I hope still, that SIP can manage more then one call through one account, if not, it is a sad thing and I have to drop the development for DECT for the PBX system at all. SIP can manage multiple simultaneous calls through one account to multiple SIP end points. But that's not what you have. As far as I can see, Asterisk is going to see multiple handsets talking (DECTing?) to a single POTS based base connected to an ATA as a single end point. Thanks in advance
[asterisk-users] sending AT commands through the SIP channel to the end device?!
Hi people! I am coding a special sollution for that I need to know if I can send AT commands in the extensions.conf, to one subscriber. Is there a way doing this through asterisk 1.6 ?! For sure anybody of you, would as why I want to do that. I want to speak to my endsystem directly with AT commands. For any advise, I would thank you kindly. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] make script 1.6.0.6 breaks up, need help!
Yes, I am installing on a 64 Bit OS... why, what does it make for a difference on what or which OS it is getting compiled?! 2009/3/22 Sebastian s...@adinet.com.uy: Are you installing on a 64bit OS?? Which Os are you using?? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: domingo, 22 de marzo de 2009 05:59 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] make script 1.6.0.6 breaks up, need help! Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 for any help and support, I would thank you people! Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.278 / Base de datos de virus: 270.11.24/2017 - Fecha de la versión: 03/22/09 17:51:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] make script 1.6.0.6 breaks up, need help!
Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 for any help and support, I would thank you people! Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building asterisk-1.6.0.6 failed!
I did the same thing, without the prefix stuff! The same error! [CC] extconf.c - extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 ta...@tux /tmp/asterisk-1.6.0.6 $ What is missing?! I am not getting smart Matt Watson schrieb: I find it a little strange that for some reason your box is using includes located in /usr/local... while there could be reason for this, that seems like a sign that something might be a little broken on your box. Also, if you don;t mind me asking... why would you want to install * directly in /usr? I could undersatnd if you are building a distribution package or something, but personaly, i would install to /usr/local or even some special place just for * just to help keep the box more organized. -- Matt On Mon, Feb 23, 2009 at 7:17 PM, Tamer Higazi th9...@googlemail.com mailto:th9...@googlemail.com wrote: Hi! I have problems building asterisk 1.6.0.6. ./configure --prefix=/usr make gets me: enerating embedded module rules ... [CC] extconf.c - extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 Now, I think this is only a dependency problem. could anyone of you tell me, which and where I am able to get the missing sources to successfully compile asterisk?! Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building asterisk-1.6.0.6 failed!
System: Gentoo Linux 2008.0, 64 Bit About package content (libsrtp 1.4.4): /* * err.h * * error status codes * * David A. McGrew * Cisco Systems, Inc. */ /* and I opested at digium bugs something: http://bugs.digium.com/view.php?id=14535 the supporter wants me to rename /usr/local = /usr/local2 what sounds for me more then strange (to take a whole partition out. Tamer Tzafrir Cohen schrieb: On Tue, Feb 24, 2009 at 09:31:21AM +0100, Tamer Higazi wrote: I did the same thing, without the prefix stuff! The same error! [CC] extconf.c - extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 ta...@tux /tmp/asterisk-1.6.0.6 $ What system is it, exactly? If Linux: what distribution? What version? If not: what OS? What version? In what package (or whatever) is /usr/local/include/err.h included? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] receiving 1st digit from a variable
Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.
Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
I would thank you for all advises.
Tamer
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Re: [asterisk-users] receiving 1st digit from a variable
exten = _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})
brings me this output:
Executing [1017649374...@officeie:1] Set(SIP/2000-007acf80,
MSNCHOICE=) in new stack
and the result is always empty!
even if I make ${EXTEN,1:3} or whaever
Danny Nicholas schrieb:
Exten = x,n,Set($NEWVAR=${EXTEN,1:1})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] receiving 1st digit from a variable
Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.
Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
I would thank you for all advises.
Tamer
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Re: [asterisk-users] receiving 1st digit from a variable
Allmost :)
It is exactly:
exten = _[0-1]X.,1,Set(MSNCHOICE=${EXTEN:0:1})
Thank you for your great support.
Tamer
Danny Nicholas schrieb:
Syntax should be
exten = _[0-1]X.,1,Set(MSNCHOICE=${EXTEN:1:1})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] receiving 1st digit from a variable
exten = _[0-1]X.,1,Set(MSNCHOICE=${EXTEN,1:1})
brings me this output:
Executing [1017649374...@officeie:1] Set(SIP/2000-007acf80,
MSNCHOICE=) in new stack
and the result is always empty!
even if I make ${EXTEN,1:3} or whaever
Danny Nicholas schrieb:
Exten = x,n,Set($NEWVAR=${EXTEN,1:1})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent: Tuesday, February 24, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] receiving 1st digit from a variable
Hi people!
I want to save the 1st letter from the ${EXTEN} variable. I don't want
to trim it, I want to RESAVE it into a new variable.
Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0
I would thank you for all advises.
Tamer
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[asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5
Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html http://127.0.0.1:8088/asterisk/static/config/cfgbasic.html I always get 404 not found! For any advises I would thank you. Here is the manager.conf and http.conf manager.conf: [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httptimeout = 60 bindaddr = 0.0.0.0 [administrator] secret = ** read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config and the http.conf: enabled=yes enablestatic=yes bindaddr=0.0.0.0 bindport=8088 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5
Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html http://127.0.0.1:8088/asterisk/static/config/cfgbasic.html I always get 404 not found! For any advises I would thank you. Here is the manager.conf and http.conf manager.conf: [general] displaysystemname = yes enabled = yes webenabled = yes port = 5038 httptimeout = 60 bindaddr = 0.0.0.0 [administrator] secret = ** read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config and the http.conf: enabled=yes enablestatic=yes bindaddr=0.0.0.0 bindport=8088 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] building asterisk-1.6.0.6 failed!
Hi! I have problems building asterisk 1.6.0.6. ./configure --prefix=/usr make gets me: enerating embedded module rules ... [CC] extconf.c - extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 Now, I think this is only a dependency problem. could anyone of you tell me, which and where I am able to get the missing sources to successfully compile asterisk?! Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem getting asterisk behind NAT to run with sipproxd
Hi people! Asterisk PBX (version 1.6.5): I have Asterisk behind a NAT (192.168.1.2) SIP Phone: A client behind NAT (192.168.1.3) Softphone: One other client somewhere in the internet (also behind an NAT). they want to speak with each other, and if they do, there is no sound. if softphone in the internet is no more behind a NAT router, it can hear SIP Phone but SIP Phone is not able to hear Softphone. siproxd is installed on the same machine where the asterisk PBX is installed, and I don't know how to configure it properly as well how the sip.conf of the asterisk PBX should have to look like. For any support I would thank you gladly. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] where to find STUN Server howto
Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Don't get asterisk to run behind NAT router
Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in general that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
