Re: [asterisk-users] Load Balancing over 2 E1 Lines
You can use the asterisk db for this. Simply set a variable to 1 or 0 if 1 set to 0 and use g2 if 0 set to 1 and use g1. - Original Message - From: Andres Jimenez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 12, 2007 11:28 AM Subject: Re: [asterisk-users] Load Balancing over 2 E1 Lines On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. That will give you the same number of calls routed to each line Is there any other possible way to make sure that all lines are used in the same amount of minutes? You are going to need an AGI app or something storing how many minutes have been routed through each line and, on every call, choosing the less used one as the line to go out. -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ --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] Realtime logic in Asterisk 1.4.16.1
Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 12:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 05:48:01 pm Mindaugas Kezys wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. were you ever able to get a solution for this? i seem the same problem when storing my sip trunks in mysql, using 1.4.16.2 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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] Realtime logic in Asterisk 1.4.16.1
On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --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] Realtime logic in Asterisk 1.4.16.1
Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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] Realtime logic in Asterisk 1.4.16.1
On Sunday 30 December 2007 06:30:09 Mindaugas Kezys wrote: Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. Please update to the latest SVN 1.4 -- this should have already been fixed. -- Tilghman ___ --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] asterisk callerid
what does your sip.conf look like for the spa
On 12/30/07, William Kenworthy [EMAIL PROTECTED] wrote:
I'm missing something simple I think:
I have an spa3102 for which I want asterisk to use the incoming pstn
callerid when it sends the call to a local extension (207).
callerid works fine for the internal phones (between each other)
The spa3102 is picking up the PSTN callerid and displays it in its own
status pages
Asterisk however, doesnt see the callerid at all.
The spa3102 is set to:
PSTN CID For VoIP CID to Yes
Dialplan 3 to (S0:207)
In the SIP messages I can see the callerid as:
From: MOBILE sip:[EMAIL PROTECTED];tag ...
To: sip:[EMAIL PROTECTED]
At the cli I get (The 935n is the user ID for the pstn)
-- Executing NoOp(SIP/Main-08169b68, 935n 207) in new
stack
-- Executing Dial(SIP/Main-08169b68, SIP/207|60|t) in new stack
Context is a basic 'catchall'
[incoming]
exten = s,1,NoOp(${CALLERID})
exten = s,n,Dial(SIP/207,90,t)
exten = s,n,Dial(SIP/202,90,t)
exten = s,n,Congestion
exten = s,n,Busy
exten = s,n,Hangup
What am I missing?
BillK
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
___
--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] Zap channels for HFC-S PCI card not responding
Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? Thanks, Jaap PS -- Below is some info regarding my configuration. === Zaptel version: 1.4.7 (incl. firmware and modules). OS: Debian etch. Loaded modules: zaphfc 13660 1 vzaphfc24984 1 zaptel185956 9 xpp,zaphfc,vzaphfc crc_ccitt 2560 1 zaptel # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) # ztcfg -vv Zaptel Version: 1.4.7-Xorcom-trunk-r5178 Echo Canceller: MG2 Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) 6 channels to configure. /etc/asterisk/zapata-channels.conf after running genzaptelconf -sd -c nl: group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default /etc/asterisk/zapata.conf (supposed to work in the Netherlands): [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Abbreviated /etc/asterisk/extensions.conf: [globals] [general] [isdn-out] exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) [internal] exten = 1000,1,Verbose(1|Extension 1000) exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() [phones] include = internal include = isdn-out Any attempts to call out result in the following CLI output: [Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION' [Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081ff9f8' not posted CLI zap show channels: Chan Extension Context Language MOH Interpret pseudodefault en default 1from-pstn en default 2from-pstn en default 4from-pstn en default 5from-pstn en default CLI zap restart: Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! Not a good idea, since that results in... CLI zap show channels: Chan Extension Context Language MOH Interpret the channels disappearing altogether! === ___ --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] Zap channels for HFC-S PCI card not responding
On Sun, Dec 30, 2007 at 04:48:39PM +0100, Jaap Winius wrote: Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? What do you mean by busy? What exactly do you see? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Realtime logic in Asterisk 1.4.16.1
Thank you! Will it come to 1.4.16.3 or 1.4.17? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Sunday, December 30, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 06:30:09 Mindaugas Kezys wrote: Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. Please update to the latest SVN 1.4 -- this should have already been fixed. -- Tilghman ___ --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] sip.conf for internetcalls.com
404 means that the number you are dialing is not available on the remote end. Is there anything that you do that makes it break or is it random ? If it is random I would speak to your ITSP. - Original Message - From: Jaap Winius [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 25, 2007 1:14 AM Subject: Re: [asterisk-users] sip.conf for internetcalls.com Quoting Justin Case [EMAIL PROTECTED]: What comes up in the Asterisk CLI? When it's not working, nothing appears in the CLI even though I've used set verbose 10. Also it can be a NAT issue? How can that lead to this intermittent behavior? I've already set nat=yes. Also, I'm using an ADSL router with a NAT; not anything like iptables. Have Asterisk register every 3-4 minutes. I'm not sure how to do that. I found defaultexpirey, but the default for it is two minutes. Anyway, why would that help with Asterisk, when my previous SIP client, a Linksys SPA3000, was configured with a register expire time of an hour and worked fine with InternetCalls.com. I think something else is going on. Using tcpdump, I see this when things are working okay: -- 23:38:05.354523 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 847 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:38:05.355065 IP bitis.umrk.to.sip 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip: SIP, length: 471 FSIP/2.0 100 Trying Via: SIP/2.0/UDP 194.221.62.198:50 . 23:38:11.007350 IP 198-rsvd-tviconnect.62.221.194.in-addr.arpa.sip bitis.umrk.to.sip: SIP, length: 507 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- The ACK packet is sent after the conversation (.) has ended. However, when it doesn't work, I see this: -- 23:42:24.736377 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 841 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via 23:42:24.736898 IP bitis.umrk.to.sip 194.120.0.198.sip: SIP, length: 445 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 194.120.0.198: 23:42:24.756967 IP 194.120.0.198.sip bitis.umrk.to.sip: SIP, length: 505 ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: S -- In this case, the ACK follows immediately after the 404 Not Found. Cheers, Jaap ___ --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] Zap channels for HFC-S PCI card not responding
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What do you mean by busy? What exactly do you see? This kind of thing: # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) Any attempts to call out result in the following CLI output: [Dec 30 16:15:41] WARNING[12918]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081ff9f8' status is 'CONGESTION' [Dec 30 16:15:41] NOTICE[12918]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/1000-081ff9f8' not posted CLI zap restart: Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Dec 30 16:32:41] ERROR[13612]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Dec 30 16:32:41] WARNING[13612]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! This and more is from my previous message (sorry, that didn't just contain configuration information). Thanks, Jaap ___ --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] asterisk callerid
sip.conf:
[iinet-PSTN]
type=friend
disallow=all
allow=ulaw
allow=g729
allow=alaw
host=dynamic
port=5061
username=iinet-PSTN
fromuser=iinet-PSTN
secret=yetta90
context=incoming
canreinvite=no
nat=route
qualify=yes
insecure=very
*Note - below I said that 9355n is the username - its actually the
display name. The username is iinet-PSTN as above. I also just changed
it to iinet-PSTN - still no callerid showing up at asterisk.
BillK
On Sun, 2007-12-30 at 10:06 -0500, C F wrote:
what does your sip.conf look like for the spa
On 12/30/07, William Kenworthy [EMAIL PROTECTED] wrote:
I'm missing something simple I think:
I have an spa3102 for which I want asterisk to use the incoming pstn
callerid when it sends the call to a local extension (207).
callerid works fine for the internal phones (between each other)
The spa3102 is picking up the PSTN callerid and displays it in its
own
status pages
Asterisk however, doesnt see the callerid at all.
The spa3102 is set to:
PSTN CID For VoIP CID to Yes
Dialplan 3 to (S0:207)
In the SIP messages I can see the callerid as:
From: MOBILE sip:[EMAIL PROTECTED];tag ...
To: sip:[EMAIL PROTECTED]
At the cli I get (The 935n is the user ID for the pstn)
-- Executing NoOp(SIP/Main-08169b68, 935n 207) in
new
stack
-- Executing Dial(SIP/Main-08169b68, SIP/207|60|t) in new
stack
Context is a basic 'catchall'
[incoming]
exten = s,1,NoOp(${CALLERID})
exten = s,n,Dial(SIP/207,90,t)
exten = s,n,Dial(SIP/202,90,t)
exten = s,n,Congestion
exten = s,n,Busy
exten = s,n,Hangup
What am I missing?
BillK
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
___
--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
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
___
--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] asterisk callerid
Fixed.
While trying to locate the fault, I had added a fixed callerid to the
extension I am using to test. Once I fixed the original problem (dont
know what!), it still wouldnt work because of the fixed callerid
Feeling a bit ... ahhh ... stupid!
Thanks all,
BillK
On Sun, 2007-12-30 at 19:49 -0500, C F wrote:
In fact the from user is making you the trouble.
On Dec 30, 2007 7:48 PM, C F [EMAIL PROTECTED] wrote:
Take out the fromuser field, you don't need it.
On Dec 30, 2007 7:37 PM, William Kenworthy [EMAIL PROTECTED] wrote:
sip.conf:
[iinet-PSTN]
type=friend
disallow=all
allow=ulaw
allow=g729
allow=alaw
host=dynamic
port=5061
username=iinet-PSTN
fromuser=iinet-PSTN
secret=yetta90
context=incoming
canreinvite=no
nat=route
qualify=yes
insecure=very
*Note - below I said that 9355n is the username - its actually the
display name. The username is iinet-PSTN as above. I also just changed
it to iinet-PSTN - still no callerid showing up at asterisk.
BillK
On Sun, 2007-12-30 at 10:06 -0500, C F wrote:
what does your sip.conf look like for the spa
On 12/30/07, William Kenworthy [EMAIL PROTECTED] wrote:
I'm missing something simple I think:
I have an spa3102 for which I want asterisk to use the incoming pstn
callerid when it sends the call to a local extension (207).
callerid works fine for the internal phones (between each other)
The spa3102 is picking up the PSTN callerid and displays it in its
own
status pages
Asterisk however, doesnt see the callerid at all.
The spa3102 is set to:
PSTN CID For VoIP CID to Yes
Dialplan 3 to (S0:207)
In the SIP messages I can see the callerid as:
From: MOBILE sip:[EMAIL PROTECTED];tag ...
To: sip:[EMAIL PROTECTED]
At the cli I get (The 935n is the user ID for the pstn)
-- Executing NoOp(SIP/Main-08169b68, 935n 207) in
new
stack
-- Executing Dial(SIP/Main-08169b68, SIP/207|60|t) in new
stack
Context is a basic 'catchall'
[incoming]
exten = s,1,NoOp(${CALLERID})
exten = s,n,Dial(SIP/207,90,t)
exten = s,n,Dial(SIP/202,90,t)
exten = s,n,Congestion
exten = s,n,Busy
exten = s,n,Hangup
What am I missing?
BillK
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
___
--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
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
--
William Kenworthy [EMAIL PROTECTED]
Home in Perth!
___
--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] app_echo.c
hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan-nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta ___ --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
