Re: [Asterisk-Users] Hangup
On Fri, 30 Jan 2004 19:22:21 -0500 Andres [EMAIL PROTECTED] wrote: Eduardo Goncalves wrote: Hi list, I'm with a little problem on my E1 (EM signaling) link. Every call a make hangs up after 2 or 3 seconds of conversation. I got the fowling messages from cli: : Zap/1-1 answered SIP/atapd-238e Urgent handler Urgent handler -- Hungup 'Zap/1-1' Urgent handler Jan 30 18:46:17 WARNING[81926]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) This is not an E1 signalling problem. Its a SIP signalling problem. You will have to get Ethereal traces of the SIP call setup to see where it is failing. Just for sure. Could it be a SIP problem, even if the chan_sip warning appears some seconds after the Hungup? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup
Hi list, I'm with a little problem on my E1 (EM signaling) link. Every call a make hangs up after 2 or 3 seconds of conversation. I got the fowling messages from cli: : Zap/1-1 answered SIP/atapd-238e Urgent handler Urgent handler -- Hungup 'Zap/1-1' Urgent handler Jan 30 18:46:17 WARNING[81926]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) Could someone help? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 support
On Thu, 22 Jan 2004 20:27:28 -0300 CW_ASN - Gus [EMAIL PROTECTED] wrote: Maybe Telefonica (the same from .ar) is not big enough! By the sight Telefónica in Brazil is not very serious, in Argentina offers ISDN in all country, for all kinds of teleservices... I'm sure of that. In Brasil, Telefonica offers ISDN, but it's a diferent comercial service (if you want voice and data in your E1), and it's more expensive. If you only want voice, the only choice is R2. Small carriers are more flexive and offers whatever singnaling you ask. regards Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 support
On Fri, 23 Jan 2004 15:49:31 -0300 CW_ASN - Gus [EMAIL PROTECTED] wrote: In Brasil, Telefonica offers ISDN, but it's a diferent comercial service (if you want voice and data in your E1), and it's more expensive. If you only want voice, the only choice is R2. Very weird, in Argentina the cost is different only for international calls in nx64; for national uses, the prices are the same for voice or data. Here, the installation and all the contract are more expensive. sucks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wink time
Hi list, I have an X100P to place some outgoing calls. But sometimes zttool shows a red alarm and after I unplug and plug the line cable, the alarm is cleared. Sometimes dialing works and sometimes not. I suspect it's a timing problem. Could someone point me on how to configure timing parameters for an X100P? thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 05 Jan 2004 15:42:25 -0600 Steven Critchfield [EMAIL PROTECTED] wrote: On my lab tests, SIP with gsm uses 26kB/s, since the link is a frame-relay and cisco routers, I've used cisco rtp header compression, and got 16kB/s per channel. Something sounds fishy here. Asterisk sends out 50 packets a second of audio(20ms). If your numbers above are per channel, you achieved a 10k reduction in 50 packets, or 204.8 bytes average per packet. Since a GSM audio packet contains 33 bytes of audio, this large header compression sounds fishy. If you are talking bits, not bytes, then it isn't that impressive. You still will probably find more efficiency in IAX. Try it and tell us your results before shooting it down. Sorry, the results are in bits per second, not bytes. my mistake. I'm doing measure tests with SIP and IAX2 trunking. I'll finish today and post the results. Thanks for the tips -- Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Trunking
Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]sip[asterisk2]PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 05 Jan 2004 10:19:24 -0700 Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote: I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking will save you more bandwidth than rtp header compression, at least if you've got multiple calls going between the two servers... I don't think it's the case. I'll have only 4 channels. On my lab tests, SIP with gsm uses 26kB/s, since the link is a frame-relay and cisco routers, I've used cisco rtp header compression, and got 16kB/s per channel. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 5 Jan 2004 11:20:08 -0600 (CST) Brian West [EMAIL PROTECTED] wrote: Why not use IAX2 trunking you can accomplish the same results with .. I tried SIP to SIP with asterisk you must do it without passwords. Because cisco doesn't compress IAX headers, only rtp. [ ]'s Eduardo On Mon, 5 Jan 2004, Eduardo Goncalves wrote: Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]sip[asterisk2]PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tor2 does not load
Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7 Did not get DONE signal. Short file maybe?? Registered Tormenta2 PCI ZT_SPANCONFIG failed on span 1: No such device or address (6) /lib/modules/2.4.18/misc/tor2.o: post-install tor2 failed /lib/modules/2.4.18/misc/tor2.o: insmod tor2 failed asterix:~# The module is listed by lsmod: asterix:~# lsmod Module Size Used byNot tainted tor2 84480 0 (unused) If I try to remove: asterix:~# rmmod tor2 Unable to handle kernel paging request at virtual address d08bc400 printing eip: d08a2c19 *pde = 0fdd4067 *pte = Oops: 0002 CPU:0 EIP:0010:[d08a2c19]Not tainted EFLAGS: 00010286 eax: d08bc000 ebx: cffe0c00 ecx: 6ea8 edx: d084cf40 esi: cef18000 edi: d08a2000 ebp: bfffecf8 esp: ce3fff48 ds: 0018 es: 0018 ss: 0018 Process rmmod (pid: 461, stackpage=ce3ff000) Stack: cffe0c00 d08b68a0 d08a2000 c01dbac4 0010 0282 c020b5dc c018ee5f cffe0c00 d08a2000 fff0 d08a4b40 d08b68a0 c020b488 0203 d08a2000 fffe ced51000 bfffecf8 c0114023 d08a2000 fff0 ced51000 bfffecf8 Call Trace: [d08b68a0] [c01dbac4] [c018ee5f] [d08a4b40] [d08b68a0] [c0114023] [c01134c7] [c0106b1b] Code: c6 80 00 04 00 00 00 8b 86 80 00 00 00 c6 80 01 04 00 00 00 Segmentation fault asterix:~# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec negotiation
Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:asterisk-ip dtmf-relay rtp-nte codec g711alaw no vad ! When I try to make a call, cisco shows codec g711alaw, but asterisk shows codec g729A (i have the licenses) and there is no audio. When I try disallow=g729, the same occurs, but this time asterisk shows codec gsm. The only way to make a call is allowing only alaw. But this is not convenience, since i need to use g279 with another endpoint (working ok). Why this negotiation problem happens? Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Debian
Hi list, Does anyone use the .deb package of asterisk? Is it stable? woks fine? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 and asterisk upgrade
Hi, We'he bought 4 g729 licenses. By the first time i installed, a ran the Registration program out of the /usr/src/asterisk directory. This worked fine, but after a few minutes, I read some docs, and reliaze that I need to run Registration from /usr/src/asterisk directory or might have problens after asterisk upgrade. So, a run registration again, this time from /usr/src/asterisk. But after asterisk upgrade, my sip calls using g729 are not working. What can I do know to solve this problem? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] D-channel
Hi list, I have an asterisk box with a TE140P using two span's (ISDN PRI euroisdn). Today, the box was running ok, but twice it's just stops to make calls and the CLI got flooded with the message D-channel on span 1 up and D-channel on span 2 up. So I stopped asterisk, unload the module, load the module, ant them started asterisk again. After this, asterisk ran ok. Could anyone give me a clue on where to look at to discover what happened? [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] D-channel down
Hi list, This morning, my asterisk box (PRI trunk) just stops to make calls. NOTICE[16401]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'Zap without any reason. Now, asterisk doesn't brings up d-channel DEBUG[11276]: File chan_zap.c, Line 6410 (pri_dchannel): Got event No event (0) on D-channel for span 1 anyone could help? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI E100P
Hi list, I have an asterisk box with a E100P using ISDN PRI: [cisco]---SIP--[asterisk]PRI[telco] Everything works fine. But /var/log/asterisk/debug is flooded with this message: Nov 14 17:17:58 DEBUG[147466]: File chan_zap.c, Line 6263 (pri_dchannel): Got event HDLC Abort (6) on D-channel for span 1 Anyone know why? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz [EMAIL PROTECTED] wrote: Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? I'd suggest your telco doing loopup and checking the circuit. My telco checked the circuit last night and didn't find anything wrong. Now I'm lost. What should I check to find what's going on? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 04 Nov 2003 22:14:17 +0800 Steve Underwood [EMAIL PROTECTED] wrote: An E1 can be a long way from the box with the right cable. However many people use the wrong cable. Using a LAN cable for an E1 often gives errors if the cable is more than just a few metres long. Although the plugs look the same, the twisted pairs should be grouped differently in an E1 cable, and it really makes a difference. If the drop cable is only a couple of metres long, a LAN cable is usually adequate. This is also true for T1s. I changed the LAN cable (about 5 meters). Now, asterisk is connected with a 1.5m cable. I hope this help. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 4 Nov 2003 09:42:36 -0600 (CST) Martin Pycko [EMAIL PROTECTED] wrote: Check if you configured the clocking from their circuit correctly. You need to have span=1,1 ... in zaptel.conf This is my zaptel.conf: span=1,1,0,cas,hdb3 alaw=1-8 em=1-8 loadzone = us defaultzone=us [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 4 Nov 2003 12:46:35 -0600 (CST) Martin Pycko [EMAIL PROTECTED] wrote: If you use TE410P make sure you have a recent zaptel from CVS. Martin My card is E400P.. About the cable lenght. The cables are like this: [telco]about 250 feet, bnc cable[baloon]---short LAN cable--[*] Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Tue, 04 Nov 2003 14:38:34 -0500 Brian D Heaton [EMAIL PROTECTED] wrote: Eduardo, Hmm, the coax is 75ohm correct? Also, since you are pushing the signal a little over 250ft you will probably need to set a different LBO value in the span= line. I'd probably try 1 or 2 (assuming a DSX-1 interface). THX/BDH Yup. 75ohm. I've tried right know and the problem occurred again. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Red Alarm
Hi list, Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm
On Mon, 3 Nov 2003 17:15:21 -0600 Don Pobanz [EMAIL PROTECTED] wrote: You also need to verify that you are using loop timing and not internal timing. (Your telco will provide timing) in zaptel.conf you should have something like span=1,1,0,ccs,hdb3,crc4 where the second 1 says to use the timing from the incoming E1 line. My zaptel.conf is like the above, except the crc4. My telco doesn't use CRC. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk -- Cisco 2620
Hi list, I'm trying to connect a cisco 2620 to my asterisk box using ISDN PRI. But I got some problems. zttool shows no alarms and Internally clocked. Asterisk starts fine, but doesn't bring up d-channel. And when I try to make a call, asterisk shows: NOTICE[16401]: File app_dial.c, Line 516 (dial_exec): Unable to create channel of type 'Zap' At the first time I configured, it worked, but after the second test call, this problems occurs. thanks Eduardo zaptel.conf: = span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = us defaultzone=us zapata.conf = [channels] context=pd switchtype=national signalling=pri_net usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel=1-15,17-31 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Starting simple switch
Hi list, I have an asterisk box with 8 zap channels (E400P, only one span, EM siginaling). And sometimes on the console, these messages apear about some channels: -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/5-1' And then: -- Hungup 'Zap/4-1' -- Hungup 'Zap/5-1' This ocurs with ramdom channels, sometimes all the channels. And sometimes, when there is active calls, the calls hangs-up. Could anyone explain why it ocurs? Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup
Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip[asterisk]EMPSTN. As endpoint I had tested another asterisk box (with a FXS), ciscoATA, cisco1750 and cisco827. The problem is the same with all. Eduardo debug when I call a PSTN number from the ATA186: == Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2210 (ast_channel_bridge): Didn't get a frame from channel: SIP/atasuporte-1413 Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2278 (ast_channel_bridge): Bridge stops bridging channels SIP/atasuporte-1413 and Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:20:02 DEBUG[1605650]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:20:04 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 debug when I hangup the ATA186 === Oct 17 19:40:25 DEBUG[278546]: File dsp.c, Line 1212 (ast_dsp_process): Requesting Hangup because the busy tone was detected on channel Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File channel.c, Line 2218 (ast_channel_bridge): Got a FRAME_CONTROL frame on channel Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File channel.c, Line 2278 (ast_channel_bridge): Bridge stops bridging channels SIP/atasuporte-4c18 and Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:40:25 DEBUG[278546]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:40:25 DEBUG[278546]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:40:25 DEBUG[81926]: File chan_sip.c, Line 544 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Oct 17 19:40:28 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT] E1 Cable pinout
Hi list, I need to connect an asterisk box to a cisco2600, using ISND PRI. My question is what cable a need. Both connectors are RJ45. thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
I did these modfications, but the problem persist. After some minutos the sip calls hang-up. :~ Eduardo On Wed, 15 Oct 2003 11:16:03 -0500 Eric Wieling [EMAIL PROTECTED] wrote: set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf Thanks for the tip. Could you explain me why these options set to yes may cause the hang up? At this time, I don't have these options in zapata.conf. What is the default? Thanks a lot Eduardo On Wed, 2003-10-15 at 09:50, Eduardo Goncalves wrote: Hi list, I have a cisco 827 with 4 fxs and an * gateway, like this: [c827]--sip-[asterisk]-em---PSTN The codec used is g711alaw over a 9Mb link. Some calls just hang up after some minutes of conversation. Cisco shows a DisconnectText=normal call clearing (16) and I found nothing in asterisk logs. Anyone can help? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD in Asterisk ?
On 15 Oct 2003 00:34:55 -0300 Juan J. Sierralta P. [EMAIL PROTECTED] wrote: I don´t have this kind of problem on my Cisco 7960 which has VAD deactivated. The problem I don't see any VAD option in AudioModes of ATA. -- Juanjo sin .sig You can disable VAD seting the AudioMode bit 0. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a00800c4b8e.html#1018462 [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip call hang up
Hi list, I have a cisco 827 with 4 fxs and an * gateway, like this: [c827]--sip-[asterisk]-em---PSTN The codec used is g711alaw over a 9Mb link. Some calls just hang up after some minutes of conversation. Cisco shows a DisconnectText=normal call clearing (16) and I found nothing in asterisk logs. Anyone can help? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
On Wed, 15 Oct 2003 11:16:03 -0500 Eric Wieling [EMAIL PROTECTED] wrote: set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf Thanks for the tip. Could you explain me why these options set to yes may cause the hang up? At this time, I don't have these options in zapata.conf. What is the default? Thanks a lot Eduardo On Wed, 2003-10-15 at 09:50, Eduardo Goncalves wrote: Hi list, I have a cisco 827 with 4 fxs and an * gateway, like this: [c827]--sip-[asterisk]-em---PSTN The codec used is g711alaw over a 9Mb link. Some calls just hang up after some minutes of conversation. Cisco shows a DisconnectText=normal call clearing (16) and I found nothing in asterisk logs. Anyone can help? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call hang up
On Wed, 15 Oct 2003 14:54:49 -0500 Eric Wieling [EMAIL PROTECTED] wrote: The default should be no. Both options listen to the audio stream. busydetect tries to determine if it hears a busy signal and if so disconnects the call. callprogress tries to determine if the call has been disconnected and disconnects the other legs of the call. Both options are buggy cause false hangups. Ok, I got it. But, could it be the cause of my problem, since the default is 'no'? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem and Fax over VoIP
On Mon, 06 Oct 2003 13:43:21 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2003-10-06 at 10:12, Eduardo Goncalves wrote: Hello, I have the fowling scenario: fxs[asterisk1]-iax-[asterisk2]e1em---PSTN If asterisk2 is your only access to the PSTN, then it doesn't make a lot of sense to do fax over VoIP. Put a couple of modems on asterisk2 with matching FXS ports and learn to use Hylafax. I can't. Because asterisk1 is at a remote site. And the only access to PSTN at this remote site is trough asterisk2. performance out of the faxing and potentially lower bills. Also it is more flexible. Not to mention that the bandwidth over VoIP to make data quality calls is around 80k of VoIP to get flakey 9.6k. Do you need to waste 10x bandwidth for sub par functionality? Is it possible to config asterisk1 to change to g.711 only when a fax transmission is detected? Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 in Brazil
On Thu, 25 Sep 2003 09:15:52 -0400 Osvaldo Mundim Junior [EMAIL PROTECTED] wrote: Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as coding. This RDSI are 30 channels completely digital which uses 128k per channel (2Mb). RDSI and ISDN are the same thing. RDSI is ISDN said in portuguese. [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Low bit rate codec (speex)
Hello, I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps. With asterisk, what's the bit rate used by speex? Is it possible to have asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to implement? thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accountcode and cdr-csv
Hello, Why does not accountcode apeer in cdr-csv? Could anyone help? ,710,01332213334,striped,710,SIP/-0810ee00,Zap/1-1,Dial,Zap/g1/01332213334,2003-08-26 10:01:59,2003-08-26 10:02:08,2003-08-26 10:02:11,12,3,ANSWERED,DOCUMENTATION ===sip.conf== [ata] type=friend context=striped host=10.0.11.160 dtmfmode=inband accountcode=pd thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo On Wed, 20 Aug 2003 11:35:14 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: VAD is evil. I hate it. I find when we used it.. you keep asking people to repeat stuff all the time.. and it was just anoying. bkw On Wed, 20 Aug 2003, WipeOut . wrote: Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman Somthing tells me that it is not supported but it is somthing that I would like to see supported as well.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VAD (silence suppression) on Asterisk
On Wed, 20 Aug 2003 13:28:59 -0500 (CDT) Brian West [EMAIL PROTECTED] wrote: Comfort Noise and VAD are diffrent things. bkw Yeap. But most devices when uses VAD looks out for gaps in speech and replaces those gaps with comfort noise. :-) [ ]'s Eduardo On Wed, 20 Aug 2003, Eduardo Goncalves wrote: When I turn on VAD on cisco ATA186, asterisk shows: Aug 20 11:12:20 NOTICE[618513]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible RCF3389 defines Payload for Comfort Noise, that is used with VAD. So I turned it off on my endpoints (ATA186 and c827-4v) Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Handled
On Wed, 13 Aug 2003 17:56:46 -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: The CNG tones are sent by the sending fax machine, not the receiving fax machine. Those tones are sent from the moment that the fax machines dials and continues until either a timeout or the receiving fax machine sends its synchronization tone. Hum, Thanks for the explanation. How is your fax machine connected to the Asterisk machine? |FAX|---|PBX|---|ATA186|SIP---|Asterisk|E1-em|PSTN| -- Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Handled
On 11 Aug 2003 15:34:38 -0500 Eric Wieling [EMAIL PROTECTED] wrote: Try adding: exten = fax,1,Dial(blah) Where Blah is the zap or SIP port your fax machine is connected to. But I want to send a fax, if I put Dial(blah), and blah is my fax machine, how could I send the fax over PSTN? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Handled
Hello, Is there any configuration in zapata.conf for fax detection (or transmission)? When I try to send a fax trought asterisk, the line 'Fax Handled:' is always set to no. The scenario is: [ata186]---sip---[asterisk]---e1 EM---[pstn] Fax sometimes goes without problem and sometimes the fax machine can't send the fax. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax Handled
On Thu, 14 Aug 2003 10:24:46 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
How is your fax machine connected to the Asterisk machine?
|FAX|---|PBX|---|ATA186|SIP---|Asterisk|E1-em|PSTN|
I suspect your problems are with the codec you're using in the SIP
connection.
I'm using G.711alaw.
My extensions.conf:
===
[globals]
TRUNK=Zap/g1
[sip]
exten = s,1,Background(demo-moreinfo)
exten = fax,1,Dial(${TRUNK}/${EXTEN})
exten = _0.,1,Dial(${TRUNK}/${EXTEN})
exten = _9.,1,Dial(${TRUNK}/${EXTEN})
Is this correct?
Eduardo
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Re: [Asterisk-Users] Fax Handled
On Mon, 11 Aug 2003 15:48:57 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
extensions.conf
===
[globals]
TRUNK=Zap/g1
[sip]
exten = _0.,1,Dial(${TRUNK}/${EXTEN})
exten = _9.,1,Dial(${TRUNK}/${EXTEN})
exten = 710,1,Dial(SIP/client1)
exten = 711,1,Dial(SIP/client2)
Okay, that's the problem. Asterisk does not detect faxes while
executing the Dial() application. You need to Background() a simple
greeting (or just 2-3 seconds of silence) before executing the Dial()
application, then provide a fax extension for Asterisk to jump to upon
detecting the CNG tones.
Could please detail how can I do this configuration?
I dial from the fax machine and then, only press the start button, when the
destination fax machine answer me and send the fax tone. So, how could asterisk dial
trought the fax exten if the fax tone are sent only when the call is completed?
--
Thanks for the help
Eduardo
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Re: [Asterisk-Users] Fax Handled
On Mon, 11 Aug 2003 15:15:08 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 11 August 2003 02:46 pm, Eduardo Goncalves wrote:
On Sun, 10 Aug 2003 01:50:33 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
Are the faxes all being sent by the same fax machine? If not,
are the faxes being detected consistently from each source? If
the remote fax machine/modem does not send CNG tones, then
Asterisk will not detect a fax.
I've tested with 3 diferent machines. Asterisk didn't detect them.
What does your dialplan look like for the s extension?
My extensions.conf are very simple. I have just one context for my trunk
connection, and the two sip endpoits.
extensions.conf
===
[globals]
TRUNK=Zap/g1
[sip]
exten = _0.,1,Dial(${TRUNK}/${EXTEN})
exten = _9.,1,Dial(${TRUNK}/${EXTEN})
exten = 710,1,Dial(SIP/client1)
exten = 711,1,Dial(SIP/client2)
Thanks
Eduardo
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Re: [Asterisk-Users] Seting up TDM40B
Martin, With KFLAGS+=-DNO_CALIBRATION uncommented, the module loads fine. Without the calibration. But I have no dial-tone on port 4. All the three other ports works fine. Could it be a hardware problem? Thanks in Advance Eduardo On Fri, 1 Aug 2003 16:11:12 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try to uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION and make clean install that should help Martin On Fri, 1 Aug 2003, Eduardo Goncalves wrote: On Fri, 1 Aug 2003 15:34:23 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: What does 'dmesg' say ? CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Timeout waiting for calibration of module 3 ProSlic died on Calibration. Module 3: Not installed Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) and then the errors that I mentioned. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seting up TDM40B
Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 15:25:49 -0500 McAughan, Matt [EMAIL PROTECTED] wrote: Have you setup the zaptel.conf and zapata.conf configuration files for how ever many ports you have on the card and then run the ztcfg -vvvc command? Since the module aren't loaded, config zaptel.conf, zapata.conf and run ztcfg will not work, right? my conf files: === zaptel.conf === fxsls=1-4 loadzone = us defaultzone=us === zapata.conf === callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=pstn signalling=fxs_ks channel=1-4 thanks []'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 15:34:23 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: What does 'dmesg' say ? CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Timeout waiting for calibration of module 3 ProSlic died on Calibration. Module 3: Not installed Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) and then the errors that I mentioned. Thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 16:11:12 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try to uncomment in zaptel/Makefile KFLAGS+=-DNO_CALIBRATION and make clean install that should help with this line uncommented, the module loads fine CSLIP: code copyright 1989 Regents of the University of California PPP generic driver version 2.4.1 Zapata Telephony Interface Registered on major 196 Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized Module 2: Initialized Module 3: Initialized Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) I also did the corrections on zapata and zaptel like James Sharp suggested. But when I try to run asterisk I get the fowling errors: == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No category context for line 1 of zapata.conf ERROR[1024]: File chan_zap.c, Line 6355 (load_module): Unable to load config zapata.conf WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! :~ Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
On Fri, 1 Aug 2003 16:31:36 -0500 (CDT) James Sharp [EMAIL PROTECTED] wrote: == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No category context for line 1 of zapata.conf zapata.conf needs to start with the line [channels] My mistake. I had commented this line when I did the regex for uncomment the others. Sorry, and thanks for the help :~ [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux flavor?
In my opinion, Debian is the best for compiling programs, because you can 'apt-get' any dependencies and its respective dependencies in a quick and clean way. You can also use auto-apt. And if you don't want to compile, you can 'apt-get install asterisk' and get asterisk running in seconds :-). But, I prefer the cvs version of asterisk. []'s Eduardo On Tue, 29 Jul 2003 15:14:37 +0200 Low, Adam [EMAIL PROTECTED] wrote: Personally, I've compiled Asterisk on Redhat and Debian without any problems on either, I think generally Asterisk compiles very easily no matter what the distro but I would recommend that you use the one you are most comfortable/experienced with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 R2 on Asterisk
On Thu, 17 Jul 2003 13:11:52 -0700 John Todd [EMAIL PROTECTED] wrote: Interestingly terse reply; perhaps you can be more specific? I have an interest in the same drivers, and there was some discussion a week ago (two weeks?) on the topic, specifically about how a driver might be written, but I heard no confirmation that there was progress or any timeframes. Anyone have any encouraging updates for those of us waiting for R2? JT I've got a libr2 from cvs. It's in alpha stage. I could't test it yet. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX G729 Codec
On 10 Jul 2003 00:29:18 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: BTW, my problems where on our private T1 line that sees round trips in the 4ms range. Our semi educated guess was that we had a problem with the jitter buffer causing echo cancel to go nutty when our ping times would occasionally jump to 20ms. When I turned off the jitter buffer, the call quality became so clear that people don't believe we are VoIP. Hi, How can I control de jitter buffer? thanks Eduado ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
On Thu, 26 Jun 2003 09:01:21 +0200 Florian Overkamp [EMAIL PROTECTED] wrote: Hi there, I have made this setup work without any special modifications. I expect it raises some strict requirements on the latency of your IP network, so that might be an issue. |cisco-ata186|sip-|asterisk|---E400P PRI -PSTN The IP network was full blast 100Mbit/s with one router inbetween. I've tested with both on the localnet (same ethernet hub) and I still get errors on the fax machine. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax and SIP
On 26 Jun 2003 12:53:40 -0400 James H. Cloos Jr. [EMAIL PROTECTED] wrote: Jim == Jim Flagg [EMAIL PROTECTED] writes: Jim Have you tried limiting your fax machines to a lower baud rate Jim like 9600. I know on Vonage this seems to help. Speaking of which, IIRC the docs for the ata mention that fax at greater than 9600 is b0rked up to a recent firmware release. You (the OP) may need to upgrade the ata to get it to work. I've tested ATA186 with a cisco827 as the H323 (or SIP) gateway and I could transmite the fax without problem. I get erros when sending faxes only when I user asterisk. :~ any tips? Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax and SIP
Hi list, I have the following scenario, and want to know what I have to do to transmit faxes trought this link: |cisco-ata186|sip-|asterisk|---EM alaw link-PSTN The codec used is g711a. When I try to transmit a fax I receive a TX FUNCTION WAS NOT COMLETED on the fax machine connected to cisco ATA186. Could someone help me? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] send DTMF digits
Hi list, What paremeter can I change to control interdigit timing? Because my PSTN provider aren't receiving all the digits I dialed on Zap/g1. My Zap/g1 are an E1 (E400P) using EM immediate sigalling. thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Only noise in zap channel
Hi list, I have an E400P using only one span with 4 channels, using EM immediate signaling. /etc/zaptel.conf span=1,1,1,cas,hdb3,yellow em=1-4 loadzone = us defaultzone=us - /etc/asterisk/zapata.conf - [channels] group = 1 context=default signalling=em channel = 1-4 This configuration works ok, I can dial on Zap/g1. But, when the other side answer the call, I only hear a lot of noise instead of the voice. Could anybody help me? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Tue, 10 Jun 2003 09:37:22 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Try in /etc/zaptel.conf to add this line: alaw=1-4 sine by default EM is used in US and the ulaw codec is being used. Martin thanks for your reply, but it still doesn't work Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Tue, 10 Jun 2003 10:08:02 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Did you do ztcfg after you added that line ? Martin yeap :~ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 and g729
On Thu, 05 Jun 2003 11:25:19 +0300 Michael Manousos [EMAIL PROTECTED] wrote: Eduardo Goncalves wrote: Hi, I have an ansterisk and a cisco 827-4v registered to a Gatekeeper. asterisk has two extensions: exten = 223,1,Dial,OH323/[EMAIL PROTECTED] Is 223 a registered alias of 827? Have you configured this destination pattern in 827? Yes and yes :~ I can call from the others cisco to this 827. I can't call from asterisk When I try to call 223 from my gnophone, 827 rings once and asterisk show the error below: *CLI ERROR[311316]: File chan_oh323.c, Line 605 (oh323_call): H323:0: Could not call 223. When I try to call my gnophone (exten 730) from the 827, asterisk show this error: *CLI assert.cxx(105) PWLib Assertion fail: Cast of NULL choice, file ../../ptclib/asner.cxx, line 3203, Error=4 Abort, Core dump, Ignore? *CLI Ignoring. Segmentation fault asterisk:~# any idea? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 and g729
On Thu, 05 Jun 2003 18:08:50 +0300 Michael Manousos [EMAIL PROTECTED] wrote: That's because you are using G.729. Don't! Try the same with G.711. Thanks Michael, using G.711 asterisk and cisco worked well... But I have to use a low bit rate codec, for use with slow links. The olny way is buyng G.279 licenses or can I use G.723? thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 and g729
Hi, I have an ansterisk and a cisco 827-4v registered to a Gatekeeper. asterisk has two extensions: exten = 223,1,Dial,OH323/[EMAIL PROTECTED] exten = 730,1,Dial(IAX/[EMAIL PROTECTED]) (IAX are working well) When I try to call each other, gnugk shows a ARJ: ARJ|10.0.11.112:1720|223:dialedDigits|730:dialedDigits|false|resourceUnavailable I think this could be a codec configuration problem. Cisco works with g729, so my oh323.conf has the configuration below: codec=G729 I have to use the 'frame=' option for this? What value? thanks in advance Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi list,
I have the follow configuration:
===
extension.conf:
===
[pstn]
ignorepat = 0
exten = _0,1,Dial(${TRUNK}/${EXTEN:1})
[default]
exten = 120,1,Dial(IAX/[EMAIL PROTECTED])
include = pstn
But, when I dial from my gnophone something like 097991269, asterisk console returns
the fallow message:
NOTICE[245775]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of
type 'Zap'
Could anyone help me?
thanks in advance
Eduardo
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Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 12:08:32 -0700 Andrew Gillham [EMAIL PROTECTED] wrote: Does it work without the group? e.g. Zap/1 Also, does 'zap show channel 1' look ok? -Andrew yeap, I tried Zap/1 and it didn't work. :~( *CLI zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Context: default Caller ID string: Destroy: 0 Signalling Type: E M Immediate Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No *CLI thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Martin The command strace -xx cat /dev/zap/1 didn't stop here my /proc/interrupts asterisk:~# cat /proc/interrupts CPU0 0: 114109 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 3:1083355 XT-PIC tor2 11: 59 XT-PIC cmpci 12: 7962 XT-PIC eth0 14: 2495 XT-PIC ide0 NMI: 0 LOC: 114078 ERR: 0 thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On 29 May 2003 14:32:01 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: What MB are you using, and what chipset is on it? Silicon Integrated Systems [SiS] 620 Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Sorry Martin, I checked the strace output and it stoped with some messages, like this: open(/dev/zap/1, O_RDONLY|O_LARGEFILE) = -1 EBUSY (Device or resource busy) Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:58:09 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: So now that I finally relize that you're using T1 or E1 circuit Do you have a ISDN PRI or an analog ciruit ? What's the status of the span in zttool or in (/proc/zaptel/1). Is it OK, RED, YELLOW ? Martin It's an E1 circuit with four channels, EM immediate signalling. I dont have ISDN neither an analog The alarm is OK Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 15:26:25 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: So it means that the board is working all right but there is problem with the telco or you're using diffrent signalling for your circuit. Martin I've just called my telephony provider and reliaze that the zaptel's signaling bits was inverted. The provider adjusted his bits and I could make a call. by the way, how can I configure the signaling bits? thanks for the help Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 16:16:29 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: I think they are hardcoded. But what do you exactly refer to by signalling bits ? Martin Bits To tell the status of a channel. It's four (ABCD) Transmit/Receive signaling bit patterns for the Idle and Seized states. You can read details here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00801123bb.shtml [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Hardware needed
I can work with digital EM - winkstart, immediate, loopstart, groundstart and ISDN with Qsig. Also with R2, but here in Brasil I prefer the first. regards Eduardo On Mon, 31 Mar 2003 15:17:45 -0600 (CST) Martin Pycko [EMAIL PROTECTED] wrote: What signalling are you going to use ? regards Martin On Mon, 31 Mar 2003, Eduardo Goncalves wrote: Hi, I'm abaut to install asterisk and I want to know if buying an E400P (Quad Span E-1 Interface) from digium my linux box will be ready (of course, after configure it) to work with PSTN and my already in use PBX (E1 interface), or I'll need to buy additional DSP's cards or whatever? thanks Eduardo Gonçalves AceNet do Brasil LTDA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
