[asterisk-users] H248 support
Hi, I am looking for H248 support in Asterisk, does anyone know if there is anything available for that? Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIY IP hardphone reference design
Hi, I am interested in building my own DIY IP hardphone to connect to Asterisk for my personal usage. Does anyone know of any good reference design in guidance me on how to build one? I am capable of building it from even raw material or circuit design if I can get some info on how to start. Thanks for all your help. Regards, Mark ___ -- 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] sound quality between two back-to-back asterisk
Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE - Asterisk - E1 - Asterisk - SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything to do with what codec to use? Thanks, Mark ___ -- 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] question about connecting with Mobile Base Station
Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark ___ -- 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] question about connecting with Mobile Base Station
Hi Andrew, Thank you for your info. I am actually looking for connecting mobile base station with asterisk via E1. Any idea on where I should start looking? Thanks, Mark On Wed, Nov 19, 2008 at 1:03 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote: On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote: Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. There are various devices. PCI GSM card, GSM to Ethernet, or the most basic is GSM to analog, then you connect it to asterisk with e.g. X100 card or SPA3000. Either the PCI or Ethernet devices should work very well -- since the call from the GSM network continues to be digital. An analog adapter will have a slower call setup time, can not support SMS or data and might have echo issues and by definition of a digital-to-analog and subsequent analog-to-digital conversion the quality of the call will be worse (but probably not noticeable). Here is one example: http://www.junghanns.net/en/GSM-PCI_produkt.html ___ -- 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] PSTN Simulator
Hi, I have Asterisk setup to run on SS7, and I would like to test it out before getting the line from my telco. Is there any testing or simulation tool that I can buy to simulate a E1/SS7 link? Could anyone give some suggestions? Thanks alot for your help in advance. Regards, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DSS1 vs SS7
Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 fax solution with Asterisk possible?
Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark ___ -- 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] Question about SS7
Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of implementing Asterisk, what kind of SS7 support is needed? Is SS7 something needs to be concerned about when using Asterisk with T1/E1? I hope someone can help me to clearify these doubts that I am having. Thanks, Mark ___ -- 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] t38modem
Hi, I am not sure if this is the right forum to ask for this. If not, let's me say sorry for my mistake in advance. Does anyone know where I can find a copy of t38modem that can work with Opal? And which Opal version should I use? Any help or hint will be greatly appreciated. Thanks, Mark ___ -- 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] PSTN to SIP
Dear all, A quick question on deploying Asterisk over E1. I am looking for a low-cost solution for bridging my E1 line and Asterisk with reasonable stability suppoing both voice and fax. Will a Digium T100 be good for that or I really need a Cisco AS 5400 for this task? What is the difference between using a Digium card vs a physical gateway server? What other alternatives are available? Your suggestions will be greatly appreciated. Thanks, Mark ___ -- 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] PSTN to SIP
Hi, Our requirement is just to be able to do voice and fax at a quality manner. What is the difference between using a physical server vs a PCI card that plugs in to the Asterisk server? Is there a big difference in terms of scalability? We are looking at a solution that can be easy-to-deploy ourselves and reasonable voice and fax quality. Thanks for your inputs. Mark ___ -- 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] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark ___ -- 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] bandwidth required for Asterisk running on T1
Hi, The T1 is 32 x 64Kbps channels ; Codec is GSM. Thank you for your suggestions. Regards, Mark On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED] wrote: mark morreny wrote: Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? What kind of T1? And what codec? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Running asterisk + T1 + ztdummy on Debian vserver
Hi I am wondering if anyone has experince running multiple instances of Asterisk using Debian vserver. The senario I want to implement is to have a couples of DIDs. Some DIDs are handled by Asterisk instance #1 and some DIDs are handled by Asterisk instance #2. The two Asterisk are within different Debian vservers. The calls are coming from a T1 line. Has anyone tried this out and maybe give me some guidelines on how that can be done? Specific questions I have are: 1. How do the two Asterisk instances connect to the T1 line via (1 or more?)ztdummy? 2. How to distribute incoming DID calls from T1 based on rules? Any suggestion will be greatly appreciated. Regards, Mark ___ -- 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] rxfax crashes Asterisk (segmentation fault)
Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [EMAIL PROTECTED]:1] Set(Zap/2-1, FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif) in new stack -- Executing [EMAIL PROTECTED]:2] RxFAX(Zap/2-1, /var/spool/asterisk-fax/1207322398.0.tif) in new st ack [Apr 4 23:20:35] NOTICE[23925]: chan_iax2.c:6025 update_registry: Restricting registration for peer ' iaxmodem' to 60 seconds (requested 50) [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Pages transferred: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size: - 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image resolution- 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Transfer Rate: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Bad rows- 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Longest bad row run - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Compression typea st_speech_unregister [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size (bytes) - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = Segmentation fault Is rxfax supposed to be working? What could have caused this problem? Thanks, Mark ___ -- 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] Strange problem with VoicemailMain
Dear all, I am having a very strange problem with VoicemailMain. When using this application to record unavail, greet, and busy, I an see the corresponding file gets created in the ../default/SIP # directory. When pressing 1 to confirm the recorded message, the *.wav file gets deleted from the file system. How can this happen? I can't figure out why. Is there any option I need to turned on to enable the recording of greeting? Thanks alot for all your help. Regards, Mark ___ -- 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] How to customize voicemail greeting
Dear friends, I am trying to configure Asterisk so that it play differnt set of voicemail greets for differnt extensions. I put my customized .wav files under the extension, but it still does not work. Asterisk still plays the default voice file. debian:/var/spool/asterisk/voicemail/default/2000# ls -al total 116 drwxr-xr-x 6 root root 4096 2008-04-01 06:03 . drwxr-xr-x 7 root root 4096 2008-03-29 01:40 .. -rw-r--r-- 1 root root 9438 2008-04-01 06:00 busy.wav -rw-r--r-- 1 root root 1815 2008-04-01 06:00 greet.wav drwxr-xr-x 2 root root 4096 2008-04-01 06:03 INBOX drwxr-xr-x 2 root root 4096 2008-03-26 09:09 Old drwxr-xr-x 2 root root 4096 2008-03-20 07:20 temp drwxr-xr-x 2 root root 4096 2008-04-01 06:03 tmp -rwxr-xr-x 1 root root 66924 2008-03-22 23:25 unavail.tmp.wav -rw-r--r-- 1 root root 2772 2008-04-01 06:02 unavail.wav Is there anything I missed out? By the way, can i use .gsm for my customized voice files? Thanks, Mark ___ -- 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 with VoiceMailMain
Dear all, I noticed a very strange problem. When I tried using VoiceMailMain to record my unavailable message, the file does not get created even though I can find the corresponding mssage from asterisk: -- SIP/2001-b6307d78 Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav, 0x82828c8 -- User ended message by pressing # -- SIP/2001-b6307d78 Playing 'auth-thankyou' (language 'en') -- SIP/2001-b6307d78 Playing 'vm-review' (language 'en') -- Saving message as is -- SIP/2001-b6307d78 Playing 'vm-msgsaved' (language 'en') -- SIP/2001-b6307d78 Playing 'vm-options' (language 'en') Also, if I copies the 'unavil.wav' inside the /2000/ directory myself, it gets deleted somehow. How come this is happening? What I want to do is to be able to copies some standard voicemail for different extension, bu so far, it does not seem to work for me yet. Can anyone give me some suggestion? Thank you very much for your kind help. Thanks, Mark ___ -- 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] breaking DNID into country code, area code, and local code
Dear friends, I am wondering if there is any efficient way of extract the country code, area code, and local code into 3 different variables from one DNID that can look like 001630233-4333 or 0086213345333? International code can be 011, or 00. National code can be 0 or 1 Country code can have 2 or 3 digits Area code can have 2 or 3 digits Local num can be 7-10 digits Is there anyway to break this down efficiently in the dialplan or AGI? Any comment will be greatly appreciated. Thanks, Mark ___ -- 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] Need help with voicemail odbc
Dear all, I am still not able to store voicemail into mysql and I am hoping someone can help me out. Here is my voicemail.cof: [general] format = wav attach = yes dbuser=ast dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages [default] ; Syntax for new entries looks like this: ; MailboxNumber = password,name,e-mail,pager,options ; (usually, the MailboxNumber is the same as the Extension) 2000 = 1234,Dave Robinson,[EMAIL PROTECTED] 2001 = 1234,Colleen Robinson,[EMAIL PROTECTED] 2002 = 1234,Matthew Robinson,[EMAIL PROTECTED] 2003 = 1234,Lisa Robinson,[EMAIL PROTECTED],,delete=yes Here is my res_odbc.conf [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. [asterisk] enabled = yes dsn = asterisk username = ast password = sqlpass pooling =no limit = o pre-connect = yes There is no error coming out of asterisk. Can anyone please tell me what could be the problem? Thanks alot for all your help. Mark ___ -- 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] Question on Dynamic Queue and Agent
Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark ___ -- 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] More info on my previous dynamic queue question
Hi, Sorry to resend the same question. This mail is just to add a few bits of details: When I tried to join the support queue, I get L RealTime: Retrieve SQL: SELECT * FROM queue_member_table WHERE interface LIKE '%' AND queue_name = 'Support' ORDER BY interface [Mar 29 10:01:52] WARNING[6203]: app_queue.c:3939 queue_exec: Unable to join queue 'Support' In show queue. it looks like the queue is set up fine: *CLI show queue Support has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: 102 (realtime) (Invalid) has taken no calls yet 101 (realtime) (Invalid) has taken no calls yet No Callers Comp-Sales has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/1001 (Unavailable) has taken no calls yet Agent/1002 (Unavailable) has taken no calls yet Agent/1003 (Unavailable) has taken no calls yet No Callers What is the problem? Is this due to the (invalide) status? I did not do the AgentCallbackLogin cuz I don't know how to get it to work with realtime queue ( there is no realtime agent ). Could anyone please help me out? Your help will be greatly appreciated. Thanks, Mark ___ -- 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] Question on Dynamic Queue and Agent
Dear Mark, I did also populate members to the queue_member_table. The output of show queue also tells me that Asterisk read the member info too. When I tried to access the queue, it saidUnable to join queue 'Support' What do you think may have gone wrong? Also, how would I be able to add a login/logoff function for the members in the queue? I could not get agentcallbacklogin to work with realtime queue. Does it work? Thank you so much for your help. Thanks, Mark On Sat, Mar 29, 2008 at 1:49 AM, Mark Michelson [EMAIL PROTECTED] wrote: mark morreny wrote: Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark There is a table for dynamic realtime queue members, called queue_members by default. If you are using Asterisk 1.4, this table should have a column for the queue to which that member belongs, the interface on which the member receives calls, the queue member's name, the member's penalty, and a boolean column for determining if the member is paused. Mark Michelson ___ -- 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] Question on Dynamic Queue and Agent
Dear Mark, Here is my queue_member table, is this how it should look? mysql SELECT * FROM queue_member_table WHERE interface LIKE '%' AND queue_name = 'Support' ORDER BY interface - ; +--+++---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--+++---+-++ |3 | 101| Support| Agent/101 |NULL | NULL | |4 | 102| Support| Agent/102 |NULL | NULL | +--+++---+-++ Many thanks, Mark On Sat, Mar 29, 2008 at 2:19 AM, mark morreny [EMAIL PROTECTED] wrote: Dear Mark, I did also populate members to the queue_member_table. The output of show queue also tells me that Asterisk read the member info too. When I tried to access the queue, it saidUnable to join queue 'Support' What do you think may have gone wrong? Also, how would I be able to add a login/logoff function for the members in the queue? I could not get agentcallbacklogin to work with realtime queue. Does it work? Thank you so much for your help. Thanks, Mark On Sat, Mar 29, 2008 at 1:49 AM, Mark Michelson [EMAIL PROTECTED] wrote: mark morreny wrote: Dear Asterisk-User friends, After realtime queues are defined, how does it work with the agents? There seems to be no db table for agents. If I can't define agents for the realtime queues in the db, how can agent login/logoff be done? Thanks alot for your help. Thanks, Mark There is a table for dynamic realtime queue members, called queue_members by default. If you are using Asterisk 1.4, this table should have a column for the queue to which that member belongs, the interface on which the member receives calls, the queue member's name, the member's penalty, and a boolean column for determining if the member is paused. Mark Michelson ___ -- 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] Asterisk not hanging up after voicemail
Hi, I am having problem with my Asterix server. It does not hand up after play the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN number; 2. After recording, I hang up and make the same call again. The first call would go through nicely with the voicemail recording, but the second call will hit a message saying the other party is busy. The only way to fix it is to reboot the Asterisk server again. Here is the CLI for the first call: -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is 8755048) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 2) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Zap/1-1, this is a voice call| not fax) in new stack -- Executing [EMAIL PROTECTED]:5] VoiceMail(Zap/1-1, 2000) in new stack -- Zap/1-1 Playing 'vm-intro' (language 'en') -- Zap/1-1 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/ZyFCyg format: wav, 0x823b2f0 What could be wrong? Thanks, Mark ___ -- 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 with socket_process: Call rejected by 127.0.0.1: Busy
Hi I am not sure why this is happening or whether it has anything to do with my iaxmodem setup. When receiving a fax via iaxmodem, I got an error message saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy* From faxstat -s, I get: JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/iaxmodem-1, we are at fax-out) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/iaxmodem-1, SIP/callwithus/0033661681) in new stack -- Called callwithus/008675533661681 -- Starting simple switch on 'Zap/1-1' -- SIP/callwithus-082370a8 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 01:54:56] NOTICE[16754]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 10) in new stack -- SIP/callwithus-082370a8 answered IAX2/iaxmodem-1 -- Redirecting Zap/1-1 to fax extension == Spawn extension (incoming, fax, 0) exited non-zero on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, getting fax) in new stack -- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, we are at fax) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(Zap/1-1, IAX2/iaxmodem) in new stack -- Called iaxmodem* [Mar 28 01:54:59] WARNING[16753]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem-4' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/1-1' == Spawn extension (fax-out, 0033661681, 2) exited non-zero on 'IAX2/iaxmodem-1' -- Hungup 'IAX2/iaxmodem-1' * Please help me. Really appreciate it. Thanks, Mark ___ -- 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] Unable to establish handshaking with fax machine
Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the future, but I am just running a single machine test right now. -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/iaxmodem-1, we are at fax-out) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/iaxmodem-1, SIP/voipuser/0033661681) in new stack -- Called voipuser/008675533661681 -- SIP/voipuser-081f99c0 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 04:01:00] NOTICE[16748]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 10) in new stack -- SIP/voipuser-081f99c0 answered IAX2/iaxmodem-1 -- Redirecting Zap/1-1 to fax extension == Spawn extension (incoming, fax, 0) exited non-zero on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, getting fax) in new stack -- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, we are at fax) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(Zap/1-1, ZAP/2) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 Thanks for helping out. I really appreciate it. Thanks, Mark ___ -- 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] Fax to DB
Hi all, I want to be able to achieve the incoming fax inside mysql along with the dailed number and CID. I understand one way to do it is to do a fax-to-email to a centralized adress and then use procmail to do the db storing. I am just wondering if there is a more straight forward why of doing that? Thanks for your input. Thanks, Mark ___ -- 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] customizing faxrcvd in PHP
Dear all, I am working on customizing hylafax's faxrcvd script into PHP. Does anyone has any sample or guideline that can share with me to give me a quick start? Two questions I have are: 1. How to simulate the receival of fax without actually sending one? 2. Where can I find the log that is echo from faxrcvd? 3. How to I config Hylafax so that it uses my PHP script instead of the original .sh script? Any help will be greatly appreciated. Thanks, Mark ___ -- 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] Getting Exec Format Error when running AGI call
Yes, it is that silly space!!! Thanks for all your help. Thanks, Mark On Mon, Mar 24, 2008 at 5:02 PM, mark morreny [EMAIL PROTECTED] wrote: Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE Here we go! 2 VERBOSE Call from - Calling phone SAY DIGITS 22 X SAY NUMBER 2233 X == Asterisk CLI == *CLI agi debug AGI Debugging Enabled *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2000-b640ba18, ) in new stack -- Executing [EMAIL PROTECTED]:2] AGI(SIP/2000-b640ba18, sample.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/sample.agi AGI Tx agi_request: sample.agi AGI Tx agi_channel: SIP/2000-b640ba18 AGI Tx agi_language: en AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1206377474.53 AGI Tx agi_callerid: 2000 AGI Tx agi_calleridname: 2000 AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 444 AGI Tx agi_rdnis: unknown AGI Tx agi_context: my-phones AGI Tx agi_extension: 444 AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx verbose Failed to execute '/var/lib/asterisk/agi-bin/sample.agi': Exec format error 2 == sample.agi: Failed to execute '/var/lib/asterisk/agi-bin/sample.agi': Exec format error AGI Tx 200 result=1 -- AGI Script sample.agi completed, returning 0 Thanks, Mark ___ -- 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] Have problem with realtime sql
Hi, I am having a strange problem with attempting to get voicemail-to-mysql to work. The biggest problem is that I am not able to store voicemail into database. So, I followed the instructor found on the web: Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install (By the way, is it necessary to update the Makefile for Asterisk 1.4.18?) After make install, I got some warning messages: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so chan_ooh323.so format_mp3.so res_config_mysql.so Is this the problem that causing Asterisk not able to store voicemessages to mysql? If so, how do I fix it? From the console, I can get realtime status ok: CLIrealtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username askuser for 1 minutes, 34 seconds. Thank you very much for your kind attentino. You help is greatly appreciated. Thanks, Mark ___ -- 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] Send received fax to different email account
Dear all, I am able to send and receive fax with Asterisk + iaxmodel + hylafax. What I want to be able to do is to 1. Stored the received fax in mysql 2. Send an email notification to he user corresponding to the incoming phone number 3. Send a SMS notification to the user's mobile phone In the hylafax setup, it seems like it can only send email to one destination email address. Is there something that can be configured? If hylafax can't do it, can anyone suggest another open source package that can accomplish the requirement? Thanks alot in advance for your help. Your help is greatly appreciated. Thanks, Mark ___ -- 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] Have problem with realtime sql
Hi Mike, Do you have any idea what is causing my voicemessages not being stored in mysql? Thanks, Mark On Wed, Mar 26, 2008 at 3:17 AM, Mike Fedyk [EMAIL PROTECTED] wrote: That's from asterisk-addons, you can ignore that error. -Original Message- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *mark morreny *Sent:* Tuesday, March 25, 2008 10:43 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Have problem with realtime sql Hi, I am having a strange problem with attempting to get voicemail-to-mysql to work. The biggest problem is that I am not able to store voicemail into database. So, I followed the instructor found on the web: Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install (By the way, is it necessary to update the Makefile for Asterisk 1.4.18?) After make install, I got some warning messages: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so chan_ooh323.so format_mp3.so res_config_mysql.so Is this the problem that causing Asterisk not able to store voicemessages to mysql? If so, how do I fix it? From the console, I can get realtime status ok: CLIrealtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username askuser for 1 minutes, 34 seconds. Thank you very much for your kind attentino. You help is greatly appreciated. Thanks, Mark ___ -- 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] Unable to obtain dialed number through ZAP
Hi all, This is not a repeated post as I am just adding more information for my previous post. Asterisk version 1.4.18 TDM card: Digium TDM411B Zaptel version 1.4.9.2 Line: PSTN line I tried to obtain the dialed number using $DNID and $CDR(DST) . All of these variable returns 's' I also tried exten = _3345335,n,Noop(this is ok) where 3345335 is my number but it does not go there. What I need to do is to try to route called based on the dialed number as I have multiple DIDs on my line. Is this something that can be done? Is this something to do with the hardware that I am using? If so, what kind of hardware do I need to accomplish this task? zapata.conf: [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no ; define channels context=incoming signalling=fxs_ks channel = 1 signalling=fxo_ks channel=2 Thanks, Mark ___ -- 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] How to capture destination number when receive call through ZAP
Dear Rob and all,
I also tried $CHANNEL before but it returned:
-- Executing [EMAIL PROTECTED]:8] NoOp(Zap/1-1, channel = Zap/1-1) in new
stack
It still does not give me the dialed number. Could you explain how to
match it again the zap channel to extract the dialed number?
Will I be able to get the dialed number if I am using a E1 line?
Thanks,
Mark
On Mon, Mar 24, 2008 at 2:29 PM, Rob Hillis [EMAIL PROTECTED] wrote:
The only method I'm familiar with for an analogue line to signal which
number was called is a very old service that loops the line first and then
dials the number. The only way to capture this would be to handle the
incoming line as a standard extension with a different context. I've only
run in to one of these services myself - and that was attached to a legacy
PABX. I seriously doubt you'd be able to order these services any more.
If this is a standard PSTN service, the only way you know which number has
been called is by matching it against the zap channel that the call has been
received on. The ${EXTEN} variable won't tell you this as you've already
found out - you'll need to examine ${CHANNEL} and match the channel to the
connected DID yourself.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
exten = s, n, Verbose(1|destination to ${EXTEN} )
${EXTEN} returns 's' instead of the actual destination number. Since I
have multiple phone numbers, I want to be able to route different calls to
different places.
Is this possible to do with Asterisk?
Thanks,
Mark
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[asterisk-users] Getting Exec Format Error when running AGI call
Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE Here we go! 2 VERBOSE Call from - Calling phone SAY DIGITS 22 X SAY NUMBER 2233 X == Asterisk CLI == *CLI agi debug AGI Debugging Enabled *CLI -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2000-b640ba18, ) in new stack -- Executing [EMAIL PROTECTED]:2] AGI(SIP/2000-b640ba18, sample.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/sample.agi AGI Tx agi_request: sample.agi AGI Tx agi_channel: SIP/2000-b640ba18 AGI Tx agi_language: en AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1206377474.53 AGI Tx agi_callerid: 2000 AGI Tx agi_calleridname: 2000 AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 444 AGI Tx agi_rdnis: unknown AGI Tx agi_context: my-phones AGI Tx agi_extension: 444 AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx verbose Failed to execute '/var/lib/asterisk/agi-bin/sample.agi': Exec format error 2 == sample.agi: Failed to execute '/var/lib/asterisk/agi-bin/sample.agi': Exec format error AGI Tx 200 result=1 -- AGI Script sample.agi completed, returning 0 Thanks, Mark ___ -- 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] Storing voicemail in mysql
Yes, I will definitely change to use a normal DB user other than root. In the meanwhile, could someone please help me out? How can I fix this problem so my voice messages can be stored in DB? Thanks, Mark On Mon, Mar 24, 2008 at 2:29 AM, Al Baker [EMAIL PROTECTED] wrote: This won't fix your problem, but it will Save you worse ones. DON'T use ROOT as your Mysql user to do this, after you get it working, set up another MySQL user, and give the account as FEW privileges as possible. mark morreny wrote: Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows no error message so I really don't know what's wrong there. -- Saved useragent wengo/v1/wengophoneng/wengo/rev12359/trunk/ for peer 2001 -- Executing [EMAIL PROTECTED]:1] VoiceMail(SIP/2001-08225788, 2000) in new stack -- SIP/2001-08225788 Playing 'vm-intro' (language 'en') -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org http://sip.voipuser.org, port 5060 -- SIP/2001-08225788 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/OayHq7 format: wav, 0x821ebd0 -- User hung up The recording is still inside a file. Is there anyway to fix it? Thanks alot for your help. Mark ___ -- 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
[asterisk-users] estimation on phone network capacity
Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent all lines are busy? We are running a support center with mostly incoming calls. Is there any rule of thumb that are typically used for this kind of estimation? Thanks, Mark ___ -- 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] Redirect and free the channel
Dear friends, I have a question about Asterisk usage. Is there anyway to configure Asterisk in such a way that once a call is forwarded out 1: to land line; or 2: to another SIP server, then the channel will be released and free up? The senarior is let's say I have an incoming ISDN line with limited channels, when all my lines are busy, I can leave one channel always open to redirect calls. For those redirect calls, the channel can be freed up again. Is this something that can be done with the existing Asterisk functionality? Thanks, Mark ___ -- 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] Unable to capture CallerID through Zap
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to obtain
the Caller ID if the calls are from the phone line.
exten = s,1,Answer()
exten = s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN}
routing to ${phonenum} )
exten = s,n, Verbose(1|callid is ${CALLID(num)})
exten = s,n,Verbose(1|callpres is ${CALLINGPRES})
exten = s,n,Dial(SIP/[EMAIL PROTECTED],60)
-- Executing [EMAIL PROTECTED]:3] Verbose(Zap/1-1, 1|incoming number is
Zap/1-1 calling to s routing to ) in new stack
-- Executing [EMAIL PROTECTED]:4] Verbose(Zap/1-1, 1| callidall is ) in
new stack
callidall is
-- Executing [EMAIL PROTECTED]:5] Verbose(Zap/1-1, 1|callid is ) in new
stack
callid is
-- Executing [EMAIL PROTECTED]:6] Verbose(Zap/1-1, 1|callpres is 0) in
new
stack
How come I am not able to get the call from ID and call to ID? Is this
something with my setup?
Thanks,
Mark
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Re: [asterisk-users] Unable to capture CallerID through Zap
Hi Gordon and all,
Thank you! It works. The other problem is that I also want to get the
destination numbers as I have multiple DIDs on the same line.
When I use exten = s,n,Noop(Destination number is ${EXTEN}) in
extentions.conf, I get s as the return value.
Is there anyway to get the actual calling number?
Thanks again,
Mark
exten = s,n,Noop(Incoming number is ${CALLERID(all)})
Note, you ought to have the number even before you Answer() the call.
Gordon
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[asterisk-users] Storing voicemail in mysql
Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows no error message so I really don't know what's wrong there. -- Saved useragent wengo/v1/wengophoneng/wengo/rev12359/trunk/ for peer 2001 -- Executing [EMAIL PROTECTED]:1] VoiceMail(SIP/2001-08225788, 2000) in new stack -- SIP/2001-08225788 Playing 'vm-intro' (language 'en') -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- SIP/2001-08225788 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/OayHq7 format: wav, 0x821ebd0 -- User hung up The recording is still inside a file. Is there anyway to fix it? Thanks alot for your help. Mark ___ -- 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] How to capture destination number when receive call through ZAP
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
exten = s, n, Verbose(1|destination to ${EXTEN} )
${EXTEN} returns 's' instead of the actual destination number. Since I have
multiple phone numbers, I want to be able to route different calls to
different places.
Is this possible to do with Asterisk?
Thanks,
Mark
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