[Asterisk-Users] Howto configure TE410P card and channels
Hi There, We install asterisk in RH9. We've T1 line and four TE410P card. We wanted to configure this card. We used ztcfg to see the channel configuration. It shows like Channels 0. How to configure the channels and the four TE410P cards. Regards SipMonsters.
RE: [Asterisk-Users] How to notify the user about new message using SMS
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
-- Executing System(SIP/192.168.0.3-0891abc8,
/scripts/sendSMS.pl -r
17083519199 -p TMOBILE -s operator -m Bartosz Wegrzyn
7734660101) in new stack
/bin/sh: line 1: /scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m Bartosz: No such file or directory
The script is not working because of the characters in caller ID.
Is there any way to change that so asterisk will pass the
variables without characters.
In addition to other responses, why don't you try passing ${CALLERIDNUM}
instead of ${CALLERID} :-)
Best regards,
Florian
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RE: [Asterisk-Users] Difficulty evaluating the return value of PlayBack (or any other extensions.conf command
I just started to play with Asterisk today and while I'm writing some IVR-like functionality in extensions.conf I would like to take a decision based on whether playing a file succeeds: Use AGI() to either check for the file presence, or to determine the rest of the dialplan logic (for example by setting a variable). That's like using an elephant to kill a mouse. Why does the documentation of Playback suggests the return value can be used intelligently when there's no obvious (documented?) way of using it? Is there a way to use it? -- Andreas Sikkema (on webmail so html post, sorry) winmail.dat
Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it
Try to comment out in your sip.conf ;qualify=yes On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -- Best regards Vlad ___ 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: FW: [Asterisk-Users] problems with asterisk and the IAX protocol
Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: Pamela, Did you resolve the problems you described? I didn't see a reply on the list but I may have missed it. -Kevin -Original Message- From: Pamela Weis [mailto:[EMAIL PROTECTED] Sent: Thursday, August 05, 2004 10:22 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] problems with asterisk and the IAX protocol Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 - asterisk1 - IAX - asterisk2 - SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone on the other side rings. But whenever I pick up the call, asterisk2 hangs up without much warning and then the telephone rings unexpectedly again and again. Here is the output of the two asterisk machines: asterisk 1: *CLI -- Accepting AUTHENTICATED call from 62.116.33.72, requested format = 256, actual format = 256 -- Executing Dial([EMAIL PROTECTED]/1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/62.116.54.194-b71d is ringing -- SIP/62.116.54.194-b71d answered [EMAIL PROTECTED]/1 == Spawn extension (local, 123, 1) exited non-zero on '[EMAIL PROTECTED]/1' -- Hungup '[EMAIL PROTECTED]/1' -- Accepting AUTHENTICATED call from 62.116.33.72, requested format = 256, actual format = 256 -- Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/62.116.54.194-6749 is ringing --- asterisk2: *CLI -- Executing Dial(SIP/-0811bef8, IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 62.116.54.194 (format G729A) -- Format for call is G729A -- IAX2[asterisk]/1 stopped sounds -- IAX2[asterisk]/1 stopped sounds -- IAX2[asterisk]/1 answered SIP/-0811bef8 Aug 5 17:57:00 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) -- Hungup 'IAX2[asterisk]/1' == Spawn extension (sip, 123, 1) exited non-zero on 'SIP/-0811bef8' Aug 5 17:57:05 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) Aug 5 17:57:06 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) -- Executing Dial(SIP/-0811bef8, IAX2/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 62.116.54.194 (format G729A) -- Format for call is G729A -- IAX2[asterisk]/2 stopped sounds -- Hungup 'IAX2[asterisk]/2' == No one is available to answer at this time I also have another question to asterisk and NAT: o) If one asterisk machine and the telephones are behind NAT, do I need a proxy to get the speech through, or should asterisk work this out on its own? Any help with my problem will be greatly appreciated. Thanks in advance. Pamela Weis ___ 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 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] res_config_odbc not working
--- Dennis Nacino [EMAIL PROTECTED] wrote: Hi, I just made it worked. The problem is brought about by the placing of odbc.ini and odbcinst.ini on /usr/local/etc. Anyway, I noticed with res_config_odbc in used, the * will only parse the first #include file it encounters. To illustrate; my extconfig.conf has the following lines lastsip.conf = odbc firstsip.conf = odbc my sip.conf has the following lines #include lastsip.conf #include firstsip.conf Once the * parses sip.conf, it will only parse #include firtsip.conf but not #include lastsip.conf. The workaround I did was to issue INSERT INTO `ast_config` ( `id` , `cat_metric` , `var_metric` , `commented` , `filename` , `category` , `var_name` , `var_val` ) VALUES ( '', '0', '0', '0', 'firstsip.conf', '', '#include ', 'lastsip.conf' ); so that * can parse whatever lastsip.conf has in the database or table. But without using res_config_odbc.so the * will parse all the #include line inside the sip.conf Any idea why * behaving like this when res_config_odbc.so is in use? -Dennis Hi, Can somebody tell me what I've been doing wrong for the res_config_odbc not to work properly. I already looked on the mailing list for some answer but I can't find one. The closes subject I found is res_odbc not working. Btw, I already followed BKW reply on that subject. I guessed the log below can tell us res_odbc.so is succefully loaded, so it easy to presumed it succesfully connected to the database. The only problem I can see is the SQL select error it encountered when it's loading sip.conf. But I have tried that sql statement on isql, PhpMyAdmin and it's succesfully executed. So, why the res_config_odbc.so is returning such error? __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ 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] e164.lu
Hello, we have set up e164.lu as a test zone, as the delegation for 2.5.3.e164.arpa hasn't been completed yet. For all those who want to call the numbers currently availble directly via SIP, please use the zone name in your enum.conf. If you decide to use the zone, please tell me at [EMAIL PROTECTED], so as soon as the 2.5.3.e164.arpa zone is ready, I will mail you, so you may disable querying e164.lu. Best Regards, Marc -- Network ManagerMarc Storck LuxAdmin.Org [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2659 0873 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] traffic termination around the globe?
hi we're a relatively new norwegian company terminating in norway. does anyone know companies that terminate traffic around the globe? we've got decent prices for .eu and .us, but we need cheaper solutions for asia, middle east and africa. regards roy ___ 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] Some Error on Asterisk....
Hello, I am New user on Asterisk.. I have some problems;; When I calledto another user from my user on soft phone, the call is correctly going, but when the other man receives the call and say "Hello", i can hear only first word and after that voice is not coming though call is going on. Also when i checked some logs i got some Warning as follows : WARNING : chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call WARNING : chan_iax2.c:5689 set_config: Ignoring port for now And i want to ask you that what is mean by this error? Transmitting (no NAT):SIP/2.0 407 Proxy Authentication Required Waiting for Positive Reply. Thanks and Regards, Nilesh Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!
Re: [Asterisk-users] Some Error on Asterisk....
Hi, It may be the problem of the CODECS that you are using in your configuration. Verify your codecs. On Mon, 09 Aug 2004 Nilesh sonavani wrote : Hello, I am New user on Asterisk.. I have some problems;; When I called to another user from my user on soft phone, the call is correctly going, but when the other man receives the call and say Hello, i can hear only first word and after that voice is not coming though call is going on. Also when i checked some logs i got some Warning as follows : WARNING : chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call WARNING : chan_iax2.c:5689 set_config: Ignoring port for now And i want to ask you that what is mean by this error? Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Waiting for Positive Reply. Thanks and Regards, Nilesh - Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage!
RE: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Robinson Tim-W10277: We are using the HFC card in point-to-point mode with DDI. I am using bri-stuff-0.0.2 as well. So, reading between the lines To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel driver and therefore must be a HFC card? Can somebody confirm this? The reason I ask is that I installed a BRI system (Single Fritz! AVM card using chan_CAPI) last week which refused to work - turned out that British Telecom had provisioned the line as a point-to-point and not point-to-multipoint as requested. Accepting that BT were going to take several days to fix their cock-up, I tried to get the card to work in point-to-point mode, but failed miserably. I could really do with getting point-to-point working on these cards as they're cheap and in plentiful supply. Have I completely misunderstood the issue and am just being stoopid, or is it not possible to do this with these cards? Many thanks, Nick. PS If HFC is the best way to go, does anybody have any recommendations on cheap HFC card suppliers in the UK? ___ 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] Sound file quality
I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers will be coming in over PSTN to a VoIP gateway and then to me by uLaw/aLaw ... would higher sampling rates gain me anything in this configuration? -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] Howto configure TE410P card and channels
On Mon, 2004-08-09 at 01:36, Snak Newyork wrote: Hi There, We install asterisk in RH9. We've T1 line and four TE410P card. We wanted to configure this card. We used ztcfg to see the channel configuration. It shows like Channels 0. How to configure the channels and the four TE410P cards. http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation Documentation exists, do your required reading before asking simple questions please. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] zaphfc - hfc pci based ISDN card : point2point DDI
What I understood from earlier discussion is that the AVM cards do not support ptp mode, only in the more expensive models. (Or was that Eicon, but those are all expensive... mmmh) ;) Either way, zaphfc/qozap seems to be the better choice for any application. Nick Barnes wrote: Robinson Tim-W10277: We are using the HFC card in point-to-point mode with DDI. I am using bri-stuff-0.0.2 as well. So, reading between the lines To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel driver and therefore must be a HFC card? Can somebody confirm this? The reason I ask is that I installed a BRI system (Single Fritz! AVM card using chan_CAPI) last week which refused to work - turned out that British Telecom had provisioned the line as a point-to-point and not point-to-multipoint as requested. Accepting that BT were going to take several days to fix their cock-up, I tried to get the card to work in point-to-point mode, but failed miserably. I could really do with getting point-to-point working on these cards as they're cheap and in plentiful supply. Have I completely misunderstood the issue and am just being stoopid, or is it not possible to do this with these cards? Many thanks, Nick. PS If HFC is the best way to go, does anybody have any recommendations on cheap HFC card suppliers in the UK? ___ 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 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] h323 direkt call instead over GK
Hi, for incomming calls, i have set an gatekkeper in h323.conf. outgoing calls wich are no sip endpoints should be sent to a h323 gateway, not the gatekeeper. If i disable the Gatekeeper, all non sip calls are routed to the Gateway. If i enable the Gatekeeper, the calls are send to the gatekeeper. here is my extensoin for the gateway. Why das asterisk send all calls to the gatekeeper instead of to the gateway? [h323-gateway] exten = _X.,1,Dial(H323/h323:[EMAIL PROTECTED]) -- Thomas Küpper 01063 Telecom GmbH Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: [EMAIL PROTECTED] E-Mail: [EMAIL PROTECTED] Homepage: http://www.01063telecom.de --- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --- ___ 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] Variables In a Context
Morning all This is probably so simple but. I have a need to set 3 values within each context ( extensions.conf ) . It would appear that this can normally only be done when an exten is called using SetVar / Global. Is this right ? Can i set these values for use at any time ? Why do i need to do this ? I have a different context for each customer , they have their own account code / userfield details. In each context i include some dialling rules dependent on their local area code when they dial i need to insert THEIR values into the db.This include is used for many different customer's Hope this makes sense. Best Regards Simon ___ 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] Some Errors on Asterisk
Hello Again, As you said It may be the problem with CODEC which i configured in my SIP.CONF. I used followoing code for CODEC in SIPCONF file : disallow=all ; Disallow all codecsallow=gsm;allow=g723.1;allow=ulaw ; Allow codecs in order of preference;allow=alaw;allow=gsm;allow=ilbc ;allow=ilbc Waiting for Positive Reply. Thanks and Regards, Nilesh == Hi,It may be the problem of the CODECS that you are using in your configuration. Verify your codecs.On Mon, 09 Aug 2004 Nilesh sonavani wrote :Hello,I am New user on Asterisk.. I have some problems;;When I called to another user from my user on soft phone, the call is correctly going, but when the other man receives the call and say "Hello", i can hear only first word and after that voice is not coming though call is going on.Also when i checked some logs i got some Warning as follows :WARNING : chan_sip.c:497 retrans_pkt: Maximum retries exceeded on callWARNING : chan_iax2.c:5689 set_config: Ignoring port for nowAnd i want to ask you that what is mean by this error?Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredWaiting for Positive Reply.Thanks and Regards,Nilesh Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers!
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel driver and therefore must be a HFC card? Can somebody confirm this? Basically yes, but ... The reason I ask is that I installed a BRI system (Single Fritz! AVM card using chan_CAPI) last week which refused to work ... if you are on kernel 2.6 and patch up your kernel with the mISDN modular isdn driver, then you might be able to run BRI P2P with mISDN+chan_capi. I could really do with getting point-to-point working on these cards as they're cheap and in plentiful supply. HFC cards are cheap as well. Check the voip-info.org wiki, as usual :-) ___ 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] Inbound not working with iconnect
Now the incoming from iConnect is working. The problem was iConnect is not taking the Contact: header at the time of registration. Though Asterisk is sending the Contact: exten@x.x.x.x header the iConnect is sending the call on phone #@x.x.x.x so all we need to do is to remove the exten in the registration line of sip.conf and use the actual phone number as the extension in extensions.conf file. Ex: Earlier we used to use register = phone #:secret@sipauth.deltathree.com/exten Now we need to use: register = phone #:secret@sipauth.deltathree.com/phone # and need to use the phone # instead of exten in the extensions.conf also. Raj --- Greg Blakely [EMAIL PROTECTED] wrote: Voicepulse is working fine, as are all my IAX devices/connections. It's inbound SIP from iConnectHere that is the problem. And they appear to be the only company (other than vonage) that has local numbers in the Minneapolis / Saint Paul metro area. So, if they don't work, I'm left holding the bag, AFAIK. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Saturday, August 07, 2004 10:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect Did you change the voicepulse context to user in you iax.conf? This is what they told me to do when I had the same problem 3 weeks ago: [voicepulse] type=user context=voicepulse-in ;auth=md5 ;secret=mysecret host=gw5.voicepulse.com qualify=yes This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Saturday, August 07, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound not working with iconnect You are right Greg, I do get a 407 error when I sniff the packets and it's the same problem now. I'm able to receive calls from Broadvoice but not from iConnect. Any help would be appreciated in this regard. Thanks, Raj Greg Blakely [EMAIL PROTECTED] wrote:This may be what I experienced in my thread New Head Appears to break SIP to iConnect. Maybe it WASN'T the fact that I upgraded my asterisk software. But, yes. I noticed the problem day before yesterday. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Raj Sent: Friday, August 06, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Inbound not working with iconnect Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj - Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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 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 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 __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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] zaphfc - hfc pci based ISDN card : point2point DDI
Holger Schurig: Basically yes, but ... Many thanks for your help - I'll stop playing with the AVM cards now! HFC cards are cheap as well. Check the voip-info.org wiki, as usual :-) Indeed. Had a look there and found a few cards, but what I really was after was recommendations for a good UK supplier or source of the cards. Cheers, Nick. ___ 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] Re: Sound file quality
In article [EMAIL PROTECTED], Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-08-09 at 06:07, David Gurr wrote: I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers will be coming in over PSTN to a VoIP gateway and then to me by uLaw/aLaw ... would higher sampling rates gain me anything in this configuration? PSTN is 8khz sample rate. So obviously a higher sample rate will not get you any where. However, 16-bit PCM-encoded 8kHz wav files would be a definite improvement over the GSM-encoded ones we currently have. I raised a feature request on bugs.digium.com (#2187) for these, and was told it will be done sometime. I hope it's soon. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] How do folks handle NAT routing?
I'm interested to hear how folks are handling NAT SIP routing issues in the wild for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to canreinvite=no in sip.conf? Any comments about real-world implementations would be welcome. Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] zaphfc - hfc pci based ISDN card : point2point DDI
On Monday 09 August 2004 14:06, Nick Barnes wrote: Holger Schurig: Basically yes, but ... Many thanks for your help - I'll stop playing with the AVM cards now! but chan_capi is in sync with * cvs, hfc-s support (bri_stuff) no -- Maurizio Marini ___ 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] sip endpoint not ringing
with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial(H323/ip$10.0.0.124:49638/18690, SIP/0699073201) in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered H323/ip$10.0.0.124:49638/18690 == Spawn extension (default, 0699073201, 1) exited non-zero on 'H323/ip$10.0.0.124:49638/18690' wit calling over pstn, the call comes from a g323 GW. on my gsm phone it rings but the sip endpoint not. whats the Problem? -- Executing Dial(H323/ip$217.9.21.6:2554/2969, SIP/0699073201) in new stack -- Called 0699073201 == No one is available to answer at this time thx -- Thomas Küpper 01063 Telecom GmbH Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: [EMAIL PROTECTED] E-Mail: [EMAIL PROTECTED] Homepage: http://www.01063telecom.de --- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --- ___ 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] asterisk-update script
On Sun, 8 Aug 2004 15:04:41 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as the Make script uses should .version file not exist. It runs well for me. That script is pretty old now, so I'm glad someone is going through and updating it. I am a bit confused by the statement ...and Mark released it as I don't know where it got released. Is it in CVS? Anyways, Steve: If you want to email me the latest changes, I will add the appropriate changes to the contributors section and update it on hacklocalhost.com (my website where the script was originally released). Thanks for taking an interest! Leif Madsen. http://www.asteriskdocs.org ___ 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] RC1 - callparking
Hi list, when I put a call in parking and take it back, I'm not able to put it again in parking. Context is empty and I receive message that extension 7 (or 70 if I'm quick) is not existing. Is this a bug or misconfiguration? Cheers -- Daniel ___ 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] asterisk with H.323 phone
Hi all, I have 'Voipac NetPhone 210' phone apparatus with H.323 support Is there any way to connect it to asterisk? What exactly I need to do. Thank you very much. ___ 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] Click to Call
Hello !! I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs I´d like to have this application in my intranet.. click on my name, than calls my number.. I´d also like to see that phone on the web... as an option How can I do that ? Is it possible to download ? Any related link ? Thanks Andrei. _ MSN Messenger: converse com os seus amigos online. http://messenger.msn.com.br ___ 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] termcapsupport not found
dirk los [EMAIL PROTECTED] wrote: I try to make an asterisk system and downloaded and unzipped the file asterisk-1.0-RC1.tar.gz. When I do the first make I got the following messages: . checking for tgetent in ltermcap...no checking for tgetent in ltinfo...no checking for tgetent in lcurses...no checking for tgetent in lncurses...no configure: error: termcapsupport not found make: *** [editline/libedit.a] Error 1 I am using a recent stable debian system Linux version 2.2.20-idepci, gcc version 2.7.2.3 Yes, Debian's definition of stable clearly means will never be updated. We run Asterisk on a Gentoo system (Linux 2.6.7-gentoo-r12 and GCC 3.3.3-r6). As an extra I installed: termcap-compat 1.2.3and evms-curses 1.00-3 In my /lib directory I see:/lib/termcap.so.2 and /lib/termcap.so.2.08 Further I see a /etc/termcap file and a /etc/terminfo file My monitor is a Compaq 1525 What can I do to get a valid termcap support? It appears that you're missing a symlink for libtermcap.so, as follows: lrwxrwxrwx 1 root root19 May 25 00:01 /lib/libtermcap.so - libtermcap.so.2.0.8 lrwxrwxrwx 1 root root19 May 25 00:01 /lib/libtermcap.so.2 - libtermcap.so.2.0.8 -rwxr-xr-x 1 root root 12124 May 21 00:55 /lib/libtermcap.so.2.0.8 -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Asterisk not starting - SOLVED!
Hello! Am Sonntag, 8. August 2004 20:12 schrieb Steven Critchfield: But * shouldn't crash with a core dump if mpg123 crashes anyway. mpg123 dumps the decoded stream to stdout (-s) and it might have some problems with id3 tags. So could it have just been that your music on hold pointed to sample-hold.mp3 and for whatever reason it wasn't available anymore and that caused your problem? Nope. Its absolutely reproducable. It's a bug in mpg123: as soon as you define a buffer with the -b option and the mp3 has an ID3 tag, mpg123 quits. If you omit the -b option it works. This has been reported to the mpg123 developer. Andi ___ 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] asterisk-update script
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 09 August 2004 09:04 am, Leif Madsen wrote: That script is pretty old now, so I'm glad someone is going through and updating it. I am a bit confused by the statement ...and Mark released it as I don't know where it got released. Is it in CVS? No, but I ended up putting it up on the WiKi. I think Mark gave it his blessings. Anyways, Steve: If you want to email me the latest changes, I will add the appropriate changes to the contributors section and update it on hacklocalhost.com (my website where the script was originally released). It's on the WiKi download page but I'll send you a copy. The version number has a sas on it as I did not know if it had continued beyond the version I had. Thanks for taking an interest! Leif Madsen. http://www.asteriskdocs.org Thank you! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBF4UfljK16xgETzkRAqLhAJwOqjAjCyZqjmBOlKsjtf3Wm/OSCQCgjYFX 18NDkGgUj+xL42K7g3ddJd8= =0O0L -END PGP SIGNATURE- ___ 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] ring tone
hi agian, i am pondering why no one ist answering to thiis problem. i found 3 list-useres who have all the same problems, but ei can not find any solution for that. wenn ich do a call to 1234 with a h323 softohone to a sip endpoint all works fine. If i make a call from PSTN to the same sip endpoint, no ring tone at the sip endpoint appears. debug output shows the same ok result like the call before. here ist the oputput from h323 client to sip: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial(H323/ip$10.0.0.124:49638/18690, SIP/0699073201) in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered H323/ip$10.0.0.124:49638/18690 == Spawn extension (default, 0699073201, 1) exited non-zero on 'H323/ip$10.0.0.124:49638/18690' and here from PSTN to sip endpoint: -- Executing Dial(H323/ip$217.9.21.6:2554/2969, SIP/0699073201) in new stack -- Called 0699073201 == No one is available to answer at this time please help if anyone has an idea to solve this. THX! -- Thomas Küpper 01063 Telecom GmbH Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: [EMAIL PROTECTED] E-Mail: [EMAIL PROTECTED] Homepage: http://www.01063telecom.de --- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --- ___ 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] iconnect inbound - so do we know how to fix it
I still have the problem, but have done a little further isolation. First, if there is no outbound iconnect section in sip.conf, my incoming calls work fine (as long as my "register" statements exist in the top section). But, when I add an outbound section, using either 'peer' or 'friend,' my incoming calls begin to fail again with the '407 Proxy Authentication' error. I've copied the section from /usr/src/asterisk/configs/sip.conf.example into my own sip.conf, and that makes no difference. Bottom line: I can have inbound or I can have outbound, but not both. One thing I've not tried is using the natrelay.deltathree.com for outbound, and sipauth.deltathree.com for inbound. Maybe that will 'fool' asterisk into thinking that they are two separate accounts. Obviously, I'm missing something here. But I've decided to lurk on the list, waiting for an answer -- with my outbound going on voicepulse, and inbound-only on iconnect. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sathya WeerasooriyaSent: Sunday, August 08, 2004 10:53 PMTo: [EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] iconnect inbound - so do we know how to fix it Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW
RE: [Asterisk-Users] How to notify the user about new message using SMS
I TRIED , SAME PROBLEM.
The value doesnot have any characters, but script fails.
Have no idea why.
Bart
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
-- Executing System(SIP/192.168.0.3-0891abc8,
/scripts/sendSMS.pl -r
17083519199 -p TMOBILE -s operator -m Bartosz Wegrzyn
7734660101) in new stack
/bin/sh: line 1: /scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m Bartosz: No such file or directory
The script is not working because of the characters in caller ID.
Is there any way to change that so asterisk will pass the
variables without characters.
In addition to other responses, why don't you try passing ${CALLERIDNUM}
instead of ${CALLERID} :-)
Best regards,
Florian
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RE: [Asterisk-Users] How to notify the user about new message using SMS
I TRIED , SAME PROBLEM.
The value doesnot have any characters, but script fails.
Have no idea why.
Bart
Hi,
-Original Message-
exten = 0,11,System(/scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m ${CALLERID}) exten = 0,12,Hangup
and this is the error:
-- Executing System(SIP/192.168.0.3-0891abc8,
/scripts/sendSMS.pl -r
17083519199 -p TMOBILE -s operator -m Bartosz Wegrzyn
7734660101) in new stack
/bin/sh: line 1: /scripts/sendSMS.pl -r 17083519199 -p
TMOBILE -s operator -m Bartosz: No such file or directory
The script is not working because of the characters in caller ID.
Is there any way to change that so asterisk will pass the
variables without characters.
In addition to other responses, why don't you try passing ${CALLERIDNUM}
instead of ${CALLERID} :-)
Best regards,
Florian
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[Asterisk-Users] uniden phones
Who are the US wholesalers selling the uniden phones? Thanks, Gary ___ 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] ChangeMonitor syntax
I'm trying to use the ChangeMonitor command on the asterisk manager API, but I can't find the syntax anywhere. Asterisk only tells me: Action: ChangeMonitor But I don't know the parameters. Can anybody help me? ___ 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] termcapsupport not found
On Mon, 2004-08-09 at 08:34, Kevin Walsh wrote: dirk los [EMAIL PROTECTED] wrote: I try to make an asterisk system and downloaded and unzipped the file asterisk-1.0-RC1.tar.gz. When I do the first make I got the following messages: . checking for tgetent in ltermcap...no checking for tgetent in ltinfo...no checking for tgetent in lcurses...no checking for tgetent in lncurses...no configure: error: termcapsupport not found make: *** [editline/libedit.a] Error 1 I am using a recent stable debian system Linux version 2.2.20-idepci, gcc version 2.7.2.3 Yes, Debian's definition of stable clearly means will never be updated. We run Asterisk on a Gentoo system (Linux 2.6.7-gentoo-r12 and GCC 3.3.3-r6). Holding my tongue on the comment from the ricer. As an extra I installed: termcap-compat 1.2.3and evms-curses 1.00-3 In my /lib directory I see:/lib/termcap.so.2 and /lib/termcap.so.2.08 Further I see a /etc/termcap file and a /etc/terminfo file My monitor is a Compaq 1525 What can I do to get a valid termcap support? It appears that you're missing a symlink for libtermcap.so, as follows: Just having the library doesn't do crap when you compiling. You need the -dev package that includes the headers to link into the library. get libncurses5-dev and all should be fine. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ChangeMonitor syntax
On Mon, 2004-08-09 at 11:24, [EMAIL PROTECTED] wrote: I'm trying to use the ChangeMonitor command on the asterisk manager API, but I can't find the syntax anywhere. Asterisk only tells me: Action: ChangeMonitor But I don't know the parameters. Can anybody help me? It takes two parameters: Channel and File. Channel is the channel that you are monitoring and want to change the filename being recorded to and File is the new filename. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Questionaire :
Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No a-) If yes creative soundblaster pci 128 is my best bet. Yes/No 2-) Which is the best codec to use with different softphones a-) SJphone : b-) Lipz4 : c-) Gnophone : 3-) Hardphones need to be plugged into the DSL router ( basically need an internet conn ) ... and need no digium cards. a-) for recieving calls : Yes/No b-) for making calls : Yes/no 4-) Sofphones run on machines different from the one running asterisk and can connect to asterisk , which routes the different phones according to contexts. Yes / no 5-) Softphone can run on the same machine as asterisk and register the same way as if they were ona different machine : yes /no 6-) Softphones can behave as independent sip phones without asterisk. Thanks niko _ Cricket maniacs ahoy! CDs, books, and more goodies! http://www.msn.co.in/Shopping/CricketShop/ Available at the cricket shop! ___ 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] Questionaire :
niko singh wrote: Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No Yes a-) If yes creative soundblaster pci 128 is my best bet. Yes/No Sorry, can't comment. 2-) Which is the best codec to use with different softphones a-) SJphone : b-) Lipz4 : c-) Gnophone : Sorry, I don't know. I would recommend trying the different ones available. 3-) Hardphones need to be plugged into the DSL router ( basically need an internet conn ) ... and need no digium cards. a-) for recieving calls : Yes/No They don't need digium cards. Depending on your network setup, you could probably get away without an internet connection for them as long as your asterisk box has connectivity. b-) for making calls : Yes/no See above answer 4-) Sofphones run on machines different from the one running asterisk and can connect to asterisk , which routes the different phones according to contexts. Yes / no yes 5-) Softphone can run on the same machine as asterisk and register the same way as if they were ona different machine : yes /no yes 6-) Softphones can behave as independent sip phones without asterisk. Yes, at least if I understand you correctly. Thanks niko -- SNIPPED-- Darren Wiebe [EMAIL PROTECTED] ___ 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] Questionaire :
On Mon, 2004-08-09 at 12:18, niko singh wrote: Hi, I have read quitea bit of the available resources and have this idea of asterisk. Would someone kindly answer these briefly 1-) Asterisk does not need a sound card...but if i am to record voice into an extension or dial from CLI ( basically use asterisk itself as a softphone ) then i need a sound card. : Yes/No Yes, but recording can be done via a phone as well. a-) If yes creative soundblaster pci 128 is my best bet. Yes/No Not necessarily. 2-) Which is the best codec to use with different softphones a-) SJphone : b-) Lipz4 : c-) Gnophone : Depends on whether you need the bandwidth or not. Gsm or ulaw work fine if you have the bandwidth. g.729 is better if you need the best in bandwidth. 3-) Hardphones need to be plugged into the DSL router ( basically need an internet conn ) ... and need no digium cards. a-) for recieving calls : Yes/No Yes b-) for making calls : Yes/no Yes 4-) Sofphones run on machines different from the one running asterisk and can connect to asterisk , which routes the different phones according to contexts. Yes / no Yes 5-) Softphone can run on the same machine as asterisk and register the same way as if they were ona different machine : yes /no They can but asterisk runs better on stand alone system. 6-) Softphones can behave as independent sip phones without asterisk. Yes. http://voip-info.org is your friend. -- respectfully, Joseph === -= Psalms 9:17 = ___ 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] Call File Routing
Here is a sample call file that I am using: MaxRetries: 2 extension: 9997 Channel: IAX2/USERID:[EMAIL PROTECTED]/14037422000 CallerID: LAKEVIEW 4037422000 Anyway, this works fine. The problem is that specifying the channel this way does not handle problems very well. If hagenhomes is down, the call will not go through. I would like to have the outgoing call routed as per one of my outgoing extensions, is this possible? I have tried to get the server to register to itself but have been unable to succeed. Any suggestions? Darren Wiebe [EMAIL PROTECTED] ___ 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] ChangeMonitor syntax
try help application changemonitor in the Asterisk CLI ___ 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] MUSIC ON HOLD PLAYING -SOLVED
I got it. My fault. I should read the instructions better. Yum installed 123mpg which does not work with asterisk. I reinstalled the old version and it is ok. Bart, Yesterday, I did update my server with some packages. After that music on hold is playing very slowly. Rest works fine. This is the list of packages I updated: 08/08/04 22:35:37 Installed: libnet10 1.0.2a-0.fdr.5.1.i386 08/08/04 22:45:21 Installed: kernel-smp 2.4.22-1.2197.nptl.i686 08/08/04 22:45:21 Installed: kernel 2.4.22-1.2197.nptl.i686 08/08/04 22:45:21 Installed: kernel-source 2.4.22-1.2197.nptl.i386 08/08/04 22:45:21 Dep Installed: php-mbstring 4.3.8-1.1.i386 08/08/04 22:45:21 Dep Installed: libmodplug 1:0.7-0.fdr.1.1.i386 08/08/04 22:45:21 Updated: php-odbc 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: php-snmp 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: php-pgsql 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: xvidcore 1.0.1-0.lvn.1.1.i386 08/08/04 22:45:21 Updated: tcl-html 8.3.5-96.0.1.i386 08/08/04 22:45:21 Updated: libpng10 1.0.15-7.i386 08/08/04 22:45:21 Updated: ethereal 0.10.5-0.1.1.i386 08/08/04 22:45:21 Updated: httpd-devel 2.0.50-1.0.i386 08/08/04 22:45:21 Updated: a52dec 0.7.4-0.lvn.7.1.i386 08/08/04 22:45:21 Updated: expectk 5.39.0-96.0.1.i386 08/08/04 22:45:21 Updated: tclx 8.3-96.0.1.i386 08/08/04 22:45:21 Updated: sox 12.17.4-4.fc1.i386 08/08/04 22:45:21 Updated: tk-devel 8.3.5-96.0.1.i386 08/08/04 22:45:21 Updated: php 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: httpd-manual 2.0.50-1.0.i386 08/08/04 22:45:21 Updated: tcl 8.3.5-96.0.1.i386 08/08/04 22:45:21 Updated: libfame 0.9.1-0.lvn.1.1.i686 08/08/04 22:45:21 Updated: libpng10-devel 1.0.15-7.i386 08/08/04 22:45:21 Updated: tcl-devel 8.3.5-96.0.1.i386 08/08/04 22:45:21 Updated: php-mysql 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: php-devel 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: gaim 1:0.81-0.FC1.i386 08/08/04 22:45:21 Updated: php-domxml 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: php-imap 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: xine-lib 1.0.0-0.lvn.0.30.rc5.1.i386 08/08/04 22:45:21 Updated: xine 0.99.2-0.lvn.2.1.i386 08/08/04 22:45:21 Updated: expect-devel 5.39.0-96.0.1.i386 08/08/04 22:45:21 Updated: tix 1:8.1.4-96.0.1.i386 08/08/04 22:45:21 Updated: apr-devel 0.9.4-2.1.i386 08/08/04 22:45:21 Updated: unrar 3.4.1-0.lvn.1.1.i386 08/08/04 22:45:21 Updated: tcllib 1.3-96.0.1.i386 08/08/04 22:45:21 Updated: mailman 3:2.1.5-6.i386 08/08/04 22:45:21 Updated: tzdata 2004b-1.fc1.noarch 08/08/04 22:45:21 Updated: flac 1.1.0-0.fdr.16.1.i386 08/08/04 22:45:21 Updated: mpeg2dec 0.4.0-0.lvn.3.b.1.i386 08/08/04 22:45:21 Updated: lame 3.96.1-0.lvn.1.1.i386 08/08/04 22:45:21 Updated: sox-devel 12.17.4-4.fc1.i386 08/08/04 22:45:21 Updated: libpostproc 1.0-0.lvn.0.13.pre5.1.i386 08/08/04 22:45:21 Updated: itcl 3.2-96.0.1.i386 08/08/04 22:45:21 Updated: php-xmlrpc 4.3.8-1.1.i386 08/08/04 22:45:21 Updated: abiword 1:2.0.1-2.i38 Does any update of those packages could brake asterisk??? I did downloaded new version of asterisk from CVS and compiled it again, but still the music on hold is like in slow motion. Any ideas ? Thanks Bart, ___ 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 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] How do folks handle NAT routing?
I'm interested to hear how folks are handling NAT SIP routing issues in the wild for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to canreinvite=no in sip.conf? Any comments about real-world implementations would be welcome. We handle it via SER with its rtpproxy/nathelper modules. Our configuration can detect automatically if the mediastream should be handled by our servers or if the endpoints successfully used STUN and can communicate directly. -- Andres Network Admin http://www.telesip.net ___ 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] uniden phones
Gary Carr wrote: James H. Thompson wrote:Who are the US wholesalers selling the uniden phones? www.thevoipconnection.com But unfortunately they are on backorder ___ 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] Sound file quality
can you use .wav files or does it have to be gsm? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 09, 2004 4:23 AM Subject: Re: [Asterisk-Users] Sound file quality On Mon, 2004-08-09 at 06:07, David Gurr wrote: I'm building a phone-in demo system to use for introducing Asterisk to prospective clients. One of the things I'm wary of is their likely preconceptions that VoIP systems will have poor audio quality. As a result, I'd like to ensure that the voice prompts I'm using have the best possible audio quality. Is it possible to use sound files at higher than 8kHz sampling? My callers will be coming in over PSTN to a VoIP gateway and then to me by uLaw/aLaw ... would higher sampling rates gain me anything in this configuration? PSTN is 8khz sample rate. So obviously a higher sample rate will not get you any where. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 My Inbox is protected by SPAMfighter 290 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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] Click to Call
Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. -g On Mon, 2004-08-09 at 11:45, Andrew Thompson wrote: Andrei Goncalves wrote: Hello !! I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs I´d like to have this application in my intranet.. click on my name, than calls my number.. I´d also like to see that phone on the web... as an option How can I do that ? Is it possible to download ? Any related link ? A couple of things: 1) It's activeX, which means it won't run outside of IE. 2) They openly state it requires features that mean it may only run in XP and ME. 3) Even after I went to IE and ran the link(I’m using XP), it wouldn’t call out for me. - Andrew Thompson http://aktzero.com/ ___ 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 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] Click to Call
Glen Hinkle wrote: Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. Did you even click the link? I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs There it is again... Try it. - Andrew Thompson http://aktzero.com/ ___ 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] Asterisk and Cisco Call Manager.
I'm in the process of doing the same thing. My approach is to declare asterisk as h323 gateway for the Cisco Call Manager, then define a route pattern to call asterisk. The strange thing that i'm dealing with now is, that the inbound RTP stream is going from the phone directly to asterisk and asterisk is sending the outbound RTP stream to asterisk. I don't know if this is a problem in asterisk or in the call manager. Salu2 Andrés Gurdeep Singh Bagga Guru escribió: Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ 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 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] ChangeMonitor syntax
On Mon, 2004-08-09 at 12:57, Holger Schurig wrote: try help application changemonitor in the Asterisk CLI I'm sure you meant show application changemonitor. That will show the dialplan application ChangeMonitor but not the Manager API ChangeMonitor command. The show manager command ChangeMonitor gives the very unhelpful help that was mentioned in the original post. I had to grep the source in order to find the proper parameters. In fact none of the show manager command ... CLI commands give any of the required parameters. Some patches would probably be in order. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Click to Call
On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote: Glen Hinkle wrote: Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. Did you even click the link? I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs There it is again... Try it. As has been mentioned here, that app is an activeX control that appears to be locked to XP and ME, and worse yet is probably not capable of being used outside of IE. It even warns you that you have to lower your security settings to get it to work. Think four or five times about the inherent danger of having a app that requires you to reduce security settings on such a piss poor security wise app. As another carrot, I would laugh at any company that wanted my to do such a thing especially if they where a computer/technology company. It would get rediculed all over any public message board I saw. As a seperate option, the CGI solution above kind of gets a similar functionality. No it doesn't use the web browser, but it would allow you to collect a phone number and issue a call out to the person requesting the call. You then could select when and how to connect. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Strange H323 problem
Hello. I have a very strange H323 problem. This is the situation: I have a Cisco 7960 phone (with IP address 10.1.1.21) connected to a Cisco CallManager (with IP address 10.1.1.10) and an Asterisk with IP address 10.1.1.22. I have managed to make the CallManager to call to asterisk using a route pattern. The strange thing is that when the call is established one RTP stream goes from 10.1.1.21 (phone) to 10.1.1.22 (asterisk) and the other goes from 10.1.1.22 (asterisk) to 10.1.1.10 (CallManager). Can somebody point me if the problem is on the asterisk side or in the CallManager side? If somebody needs it, I have the complete ethereal session in http://www.totexa.cl/ccm_session Thanks. Andrés ___ 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] iconnect inbound - so do we know how to fix it
Raj, yes your post helped me. Just to complete the whole thing and clarify the problem that was posted by Greg Blakely; First, if there is no outbound iconnect section in sip.conf, my incoming calls work fine (as long as my register statements exist in the top section). But, when I add an outbound section, using either 'peer' or 'friend,' my incoming calls begin to fail again with the '407 Proxy Authentication' error. When there is a context created in SIP.CONF for iconnect outgoing, we should point it correctly to extensions.conf. Reason is now the incoming too land in this context. Thanks Sathya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raj Sent: Monday, August 09, 2004 5:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iconnect inbound - so do we know how to fix it May be you can find the solution in my post: http://lists.digium.com/pipermail/asterisk-users/2004-August/058014.html Raj --- Vladyslav [EMAIL PROTECTED] wrote: Try to comment out in your sip.conf ;qualify=yes On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -- Best regards Vlad ___ 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 __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ 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] Click to Call
Steven Critchfield wrote: . As a seperate option, the CGI solution above kind of gets a similar functionality. No it doesn't use the web browser, but it would allow you to collect a phone number and issue a call out to the person requesting the call. You then could select when and how to connect. One of my students wrote an app that integrates TACI, which I got via a link on the asterisk list sometime back, and Postgres, with a front-end that allows click to call capability. I have contacted him to see if he would like to clean it up a bit and release it. TACI is at http://www.azxws.com/asterisk/ I don't know if it's being actively developed/maintained. B. ___ 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] Fedora FC2 and Zaptel (Torisa)
Followed the instructions on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format FATAL: Error inserting tor2 (/lib/modules/2.6.5-1.358smp/misc/tor2.ko): Invalid module format FATAL: Error running install command for tor2 A dmesg shows this: tor2: version magic '2.6.5-1.358custom SMP 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.358smp SMP 686 REGPARM 4KSTACKS gcc-3.3' ___ 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] inbound/outbound trunk groups
Is it possible to limit which Zap channels answer the phone? If I have four numbers but only want asterisk to answer the 1st channel, and allow all four channels out bound in a hunt group. I think this is called a trunk group, does it support this? Any information would be helpful. [EMAIL PROTECTED] ___ 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] Re: Grandstream Message Waiting light
And here I was trying to figure out how to kill the blinking display :-) OK - dumb newbie award hereby rewarded to me. Thanks. And I had already checked the wiki and done what you suggested in sip.conf - so my stupidity wasn't total :-) Stupidity may be a bit strong in any case. The real stupidity was in not putting a LED under the message button in the first place. Then we could assume the intuitively obvious and wouldn't need to be confused about the multiple meaning of the flashing display. Steve Besch ___ 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] Click to Call
Steven Critchfield wrote: On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote: Glen Hinkle wrote: Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. snip As a seperate option, the CGI solution above kind of gets a similar functionality. No it doesn't use the web browser, but it would allow you to collect a phone number and issue a call out to the person requesting the call. You then could select when and how to connect. Ok, in that scenario I can see how the call file would provide similar functionality. I apologize to Glen Hinkle, and to the list, for my comment. - Andrew Thompson http://aktzero.com/ ___ 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] Fedora FC2 and Zaptel (Torisa)
Hello Matt, I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver Matt Schulte wrote: Followed the instructions on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format FATAL: Error inserting tor2 (/lib/modules/2.6.5-1.358smp/misc/tor2.ko): Invalid module format FATAL: Error running install command for tor2 A dmesg shows this: tor2: version magic '2.6.5-1.358custom SMP 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.358smp SMP 686 REGPARM 4KSTACKS gcc-3.3' ___ 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 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] Re: Sound file quality
In article [EMAIL PROTECTED], hank [EMAIL PROTECTED] wrote: can you use .wav files or does it have to be gsm? You can use .wav files. They should be PCM format, 8000Hz sampling, 16 bit mono. Windows Sound Recorder can produce them, as can sox. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Call File Routing
Thank you very much Jeremy. This works perfectly. I have been struggling for a long time to get this to work but it now works perfectly. I have added a comment to the wiki: http://www.voip-info.org/wiki-Asterisk+auto-dial+out#comments Thanks Again, Darren Wiebe [EMAIL PROTECTED] Jeremy Hall wrote: Aleph Communications scribbled on Monday, August 09, 2004 10:50 AM: Here is a sample call file that I am using: MaxRetries: 2 extension: 9997 Channel: IAX2/USERID:[EMAIL PROTECTED]/14037422000 CallerID: LAKEVIEW 4037422000 Anyway, this works fine. The problem is that specifying the channel this way does not handle problems very well. If hagenhomes is down, the call will not go through. I would like to have the outgoing call routed as per one of my outgoing extensions, is this possible? I have tried to get the server to register to itself but have been unable to succeed. Any suggestions? I have not yet tried it, but earlier today I had a similar question and Twisted in IRC gave me the following response: twisted Channel: Local/[EMAIL PROTECTED] twisted Application: Dial twisted and then the arguments go in the appropriate place Hope that helps. Jeremy Disclaimer: 9/8/2004 MPC Computers is providing the following information in compliance with federal regulations: MPC Computers, LLC 906 E. Karcher Road Nampa, Idaho 83687 1-888-224-4247 http://www.mpccorp.com To discontinue receiving e-mail communications from MPC in the future, please go to: http://www.mpccorp.com/email/manage.html and follow the instructions. ___ 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 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] Sound file quality
hank wrote on 8/9/04, 12:10 PM: can you use .wav files or does it have to be gsm? See the wiki http://www.voip-info.org/tiki-index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk ___ 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] uniden phones
Actually, they are in stock now. At least they were able to fill waiting orders. Mine came in today. -Nate -Original Message- From: Paul Zimm [mailto:[EMAIL PROTECTED] Sent: Monday, August 09, 2004 12:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] uniden phones Gary Carr wrote: James H. Thompson wrote:Who are the US wholesalers selling the uniden phones? www.thevoipconnection.com But unfortunately they are on backorder ___ 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 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] AbsoluteTimeout Inside A Macro
Hi all,
Is it just me and not reading the docs right, or has anybody else had
problems with the AbsoluteTimeout application and the 'T' extension when
used inside a macro?
[macro-attended]
; ARG1 is the device to dial out on, SIP or Zap, or whatever
; ARG2 is the extension to dial using 'attended' dialing
exten = s,1,AbsoluteTimeout(30)
exten = s,2,AGI(attended-extension,${ARG1},${ARG2})
; attended-extension takes a device string and an extension
; and builds a dial string according to some crazy internal logic
exten = s,3,Dial(${DIALSTRING},5,t)
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Goto(s,1)
exten = T,1,NoOp(i got here here)
exten = T,2,Goto(s,1)
The purpose of this macro is to be able to say something like
exten = _8XX,1,Macro(attended,SIP,${EXTEN})
and have the the dialed extension rung, then, if no answer within 5
seconds, have the dialed extension plus an 'attendant' for that
extension rung, (etc. etc. etc.). If nobody answers after 30 seconds,
the caller is (read 'will be') offered the chance to leave a voicemail,
otherwise re-enter the loop, ringing the 'full' attendant list for the
requested extension.
When I test this, everything works according to plan, except when
AbsoluteTimeout expires, my T extension inside the macro is not
executed, the call is simply hungup. What am I doing wrong?
Thanks,
Chris
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RE: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)
I did that, now I get this error on compile: make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/lib/modules/2.6.5-1.358' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/lib/modules/2.6.5-1.358' make: *** [linux26] Error 2 Could this be from following directions on voip-info ?? Thanks again, Matt -Original Message- From: Oliver [mailto:[EMAIL PROTECTED] Sent: Monday, August 09, 2004 2:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa) Hello Matt, I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver Matt Schulte wrote: Followed the instructions on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format FATAL: Error inserting tor2 (/lib/modules/2.6.5-1.358smp/misc/tor2.ko): Invalid module format FATAL: Error running install command for tor2 A dmesg shows this: tor2: version magic '2.6.5-1.358custom SMP 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.358smp SMP 686 REGPARM 4KSTACKS gcc-3.3' ___ 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 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 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] Click to Call
Andrei, It is an activeX control and it no longer needs to have the security level changed. (The earlier version was an unsigned applet that required the security level change until I bought the code signing certificate early this year.) If you want to have a soft phone running in a web browser, your approaches are somewhat limited and an ActiveX control is probably the most expedient approach for the general user population. (read: very few people from this list ;-) ) So, if you're Intranet can reach the Internet, connect your Asterisk server to FWD and simply have the users' click on your icon and call your FWD account. This HTML code snippet does it: A HREF=http://fwd.pulver.com/callme.php?userid=5; IMG SRC=http://fwd.pulver.com/myicon.php?userid=5; FWD# 5 /A Change it to use your userid, and you may replace the image as desired. If you simply want click dial to dial, your choices are many... On asterisk, I use the create a file in /var/spool/asterisk/outgoing approach with great success. Here's a simple shell script I use; it could easily be moved to php or a CGI after cleansing the parameters. (There must be complete examples on voip-info.org) #!/bin/bash # # usage: callme fromnumber tonumber # PHONE=SIP/$1 DEST=$2 TMPFILE=/tmp/$$.call cat $TMPFILE XXX # Channel: $PHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # Context: internal-origination Extension: $DEST Priority: 1 XXX mv $TMPFILE /var/spool/asterisk/outgoing/ # # end Best Regards, Ed Guy @ pulver.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, August 09, 2004 2:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Click to Call On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote: Glen Hinkle wrote: Just write a CGI script that places a file in in the outgoing calls directory. /var/spool/asterisk/outgoing, I believe. This will accomplish what you're wanting. Did you even click the link? I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs There it is again... Try it. As has been mentioned here, that app is an activeX control that appears to be locked to XP and ME, and worse yet is probably not capable of being used outside of IE. It even warns you that you have to lower your security settings to get it to work. Think four or five times about the inherent danger of having a app that requires you to reduce security settings on such a piss poor security wise app. As another carrot, I would laugh at any company that wanted my to do such a thing especially if they where a computer/technology company. It would get rediculed all over any public message board I saw. As a seperate option, the CGI solution above kind of gets a similar functionality. No it doesn't use the web browser, but it would allow you to collect a phone number and issue a call out to the person requesting the call. You then could select when and how to connect. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 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] quadBRI + FAX
Can anybody suggest how to setup company having 3 bri connections to local telco. As far as I understood iax is supper good for interconeting branch offices. quadBRI is the best solution if you have more than 1 ISDN channel. But every company still needs a fax. More or less hylafax is suitable for this. Any analog voice, or capi aproache to hylafax is also suitable. 1) If you use Zaptel interface quadBRI card, what to do with faxes ? You can put cisco ATA-186(sip) after asterisk, and then an ordinary modem but in this case you lose functionality, quality and so on. 2)Another aproche is to use mISDN + hfc_multiboard + chan_capi, and capifaxrecv ... in that case you need to start a lot of threeds to listen on each BRI interface for incomming faxes. Futher more mISDN is totaly unstable with fritz, mayby with hfc it is stable? 3) app_fax_recv, but stil no integration with hylafax ( not sure ) hard to compile... lots of limitations Any working and stable examples are wellcome. ___ 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] Re: Grandstream Message Waiting light
I agree with you there, I wouldn't feel too stupid, the same thing happened to me when I purchased my BT101... The picture on the website and on the box shows the message button lit up in red, I naturally assumed that when MWI was triggered, that would happen... but I quickly realized that it was the blinking display and not the button when I took the faceplate off and saw that there's no LED under there :) -Chris - Original Message - From: Stephen R. Besch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 09, 2004 12:25 PM Subject: [Asterisk-Users] Re: Grandstream Message Waiting light And here I was trying to figure out how to kill the blinking display :-) OK - dumb newbie award hereby rewarded to me. Thanks. And I had already checked the wiki and done what you suggested in sip.conf - so my stupidity wasn't total :-) Stupidity may be a bit strong in any case. The real stupidity was in not putting a LED under the message button in the first place. Then we could assume the intuitively obvious and wouldn't need to be confused about the multiple meaning of the flashing display. Steve Besch ___ 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 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] Fedora FC2 and Zaptel (Torisa)
No, i made a mistake ... the symlink is actually linux-2.6 - /lib/modules/2.6.5-1.358/build I forgot the build - I am very sorry about that. Matt Schulte wrote: I did that, now I get this error on compile: make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/lib/modules/2.6.5-1.358' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/lib/modules/2.6.5-1.358' make: *** [linux26] Error 2 Could this be from following directions on voip-info ?? Thanks again, Matt -Original Message- From: Oliver [mailto:[EMAIL PROTECTED] Sent: Monday, August 09, 2004 2:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa) Hello Matt, I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver Matt Schulte wrote: Followed the instructions on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format FATAL: Error inserting tor2 (/lib/modules/2.6.5-1.358smp/misc/tor2.ko): Invalid module format FATAL: Error running install command for tor2 A dmesg shows this: tor2: version magic '2.6.5-1.358custom SMP 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.358smp SMP 686 REGPARM 4KSTACKS gcc-3.3' ___ 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 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 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 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] H323 under asterisk RC1 ?
This evening I tried to install asterisk RC1 .rpm on a Fedora box... but how can I re-add openh323 support? or does it contain an alternate h323 support? thanks in advance ___ 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] quadBRI + FAX
use the fourth bri span with a BRI TA with 2 FXS.. 3Com/USR has a working model, but there are plenty more. [EMAIL PROTECTED] wrote: Can anybody suggest how to setup company having 3 bri connections to local telco. As far as I understood iax is supper good for interconeting branch offices. quadBRI is the best solution if you have more than 1 ISDN channel. But every company still needs a fax. More or less hylafax is suitable for this. Any analog voice, or capi aproache to hylafax is also suitable. 1) If you use Zaptel interface quadBRI card, what to do with faxes ? You can put cisco ATA-186(sip) after asterisk, and then an ordinary modem but in this case you lose functionality, quality and so on. 2)Another aproche is to use mISDN + hfc_multiboard + chan_capi, and capifaxrecv ... in that case you need to start a lot of threeds to listen on each BRI interface for incomming faxes. Futher more mISDN is totaly unstable with fritz, mayby with hfc it is stable? 3) app_fax_recv, but stil no integration with hylafax ( not sure ) hard to compile... lots of limitations Any working and stable examples are wellcome. ___ 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 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] Application asterisk uses obsolete OSS audio interface
Should I be concerned about this? It seems to only happen when my MoH switches songs. The songs sound as good as an 8k/s song would. Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED]
[Asterisk-Users] called and callers buttons on bt100
is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami ___ 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] Snom Intercom
I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and trying to use the following dialplan: exten = 5100,1,SipAddHeader(Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris,,v) or exten = 5100,1,SipAddHeader(Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris) or exten = 5100,1,SetVar(_SIPADDHEADER=Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris,,v) None of the above work. Can somebody tell me that this either is or is not possible? Ryan ___ 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] Inbound Call Errors...
I have searched all over the web and have not really found anything
related to this error The only thing I found is related to a
system stops responding on DTMF, which does not happen here... THe
following is the output from the CLI:
*CLI 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc:
Allocating new SIP call for
[EMAIL PROTECTED]
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:6991 handle_request:
Check for res for
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:1605 update_user_counter:
is not a local user
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:4423 build_route:
build_route: Contact hop: sip:65.67.76.30:5060
2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1255 pbx_extension_helper:
Launching 'Congestion'
2004-08-09 17:36:29 DEBUG[245775]: channel.c:652 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/65.67.76.30-0814e4f0'
2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1827 ast_pbx_run: Spawn
extension (bogon-calls,5462000,1) exited non-zero on
'SIP/65.67.76.30-0814e4f0'
2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1255 pbx_extension_helper:
Launching 'Congestion'
2004-08-09 17:36:29 DEBUG[245775]: pbx.c:1947 ast_pbx_run: Spawn
extension (bogon-calls,h,1) exited non-zero on
'SIP/65.67.76.30-0814e4f0'
2004-08-09 17:36:29 DEBUG[245775]: cdr_pgsql.c:100 pgsql_log:
cdr_pgsql: inserting a CDR record.
2004-08-09 17:36:29 DEBUG[245775]: cdr_pgsql.c:103 pgsql_log:
cdr_pgsql: SQL command executed: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode,uniqueid,userfield)
VALUES ('2004-08-09
17:36:29','65.67.76.30','65677630','5462000','bogon-calls',
'SIP/65.67.76.30-0814e4f0','','Congestion','',0,0,'NO
ANSWER',3,'','1092090989.0','')
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed
to grab lock, trying again...
2004-08-09 17:36:29 DEBUG[245775]: channel.c:733 ast_hangup: Hanging up
channel 'SIP/65.67.76.30-0814e4f0'
2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1717 sip_hangup:
sip_hangup(SIP/65.67.76.30-0814e4f0)
2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1732 sip_hangup:
update_user_counter() - decrement inUse counter
2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1605 update_user_counter:
is not a local user
2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of
Response 101: Not Found
2004-08-09 17:37:10 DEBUG[229390]: chan_sip.c:2332 sip_alloc:
Allocating new SIP call for
[EMAIL PROTECTED]
I can make the calls from phone to phone and this same access server is
sending and receiving calls to and from the existing call manager that
I am working to eliminate just fine. Can someone give me any ideas as
to the root of this issue.
Stephen Malenshek
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Re: [Asterisk-Users] h323 direkt call instead over GK
Thomas Kuepper wrote: Hi, for incomming calls, i have set an gatekkeper in h323.conf. outgoing calls wich are no sip endpoints should be sent to a h323 gateway, not the gatekeeper. If i disable the Gatekeeper, all non sip calls are routed to the Gateway. If i enable the Gatekeeper, the calls are send to the gatekeeper. here is my extensoin for the gateway. Why das asterisk send all calls to the gatekeeper instead of to the gateway? [h323-gateway] exten = _X.,1,Dial(H323/h323:[EMAIL PROTECTED]) That is not a correct H323 exten line H323/[EMAIL PROTECTED] Jeremy McNamara ___ 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] CVS download
I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S- server_register(fpm-world-mix.mp3, 1.1, , , , , ) S- Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:[EMAIL PROTECTED] ___ 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] Snom Intercom
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 09 August 2004 05:58 pm, rayers.list wrote: I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. Did you read the Snom admin/technical manual? I'm pretty sure I saw there how to do it. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBGA1VljK16xgETzkRAhWBAKCfMWPPpqzsoS5gr8Yl0xuGc0u+UQCgnlUp qE5tNBLx/o+tR16tjq4EuUU= =lHX7 -END PGP SIGNATURE- ___ 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] Fedora FC2 and Zaptel (Torisa)
Or you can install the kernel 2.6.7 and all those little worries disappeared. I don't know what they did in FC2 to get it so wrong with their kernel... Jean-Yves On 10/08/2004, at 5:37 AM, Oliver wrote: I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ 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] Questionaire :
On 9 Aug 2004 at 12:35, Joseph wrote: respectfully, Joseph === -= Psalms 9:17 = Woah! The wicked shall be turned into hell, and all the nations that forget God. Bit intense for an asterisk mailing list! :-) Matt Riddell ___ 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] Fedora FC2 and Zaptel (Torisa)
I just ended up reverting to 2.4.22-1.2188.nptl. Nothing really all that interesting in 2.6.x for a production server yet, for me anyway. -Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Yves Avenard Sent: Monday, August 09, 2004 8:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa) Or you can install the kernel 2.6.7 and all those little worries disappeared. I don't know what they did in FC2 to get it so wrong with their kernel... Jean-Yves On 10/08/2004, at 5:37 AM, Oliver wrote: I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ 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 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] Sound file quality
David == David Gurr [EMAIL PROTECTED] writes: David As a result, I'd like to ensure that the voice prompts I'm David using have the best possible audio quality. David My callers will be coming in over PSTN to a VoIP gateway and David then to me by uLaw/aLaw ... The optimal quality in the case where pstn is involved would be from: Using pro-quality (which these days does not necessarily mean pro $$$) recording equipment Recording in DAT quality (16 bit 48 kHz) the recording equipment is more likely to support that than 16 kHz or 32 kHz 32bit float rather than 16 bit int is ok, too Edit the files at this point for lead time, trail time, equal volume, et al. Use a high quality resampling algorithm (in sox use polyphase) to resample to signed-16bit 8 kHz. Optionally use a band-pass filter here to drop stuff outside of the PSTN frequenc range. If you only do one of alaw/ulaw, you might as well convert the files to that, else leave them as signed-16bit You can still get things like phase distortion if the path has jitter and the receiver does not jitter-buffer. You will also need to do some experimenting to determine the optimal amplitude to avoid both clipping and too-little use of the available u/a-law bandwidth. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] ___ 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] Snom Intercom
Yes, I read that. It is not in there. It does mention setting the Auto-Answer for the phone. However, I want an intercom, I don't want a door phone. The Auto-Answer feature just sets it so it answers all calls automatically. Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Szmidt Sent: Monday, August 09, 2004 6:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Snom Intercom -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 09 August 2004 05:58 pm, rayers.list wrote: I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. Did you read the Snom admin/technical manual? I'm pretty sure I saw there how to do it. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBGA1VljK16xgETzkRAhWBAKCfMWPPpqzsoS5gr8Yl0xuGc0u+UQCgnlUp qE5tNBLx/o+tR16tjq4EuUU= =lHX7 -END PGP SIGNATURE- ___ 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 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] Answer Call Waiting from Call Forward to Cell Phone
Hello- I am new to the list, so forgive me if this has been answered, but I haven't seen it on google yet. What I want to know is, is there any way to send a hook-flash signal from a cell phone (and then have Asterisk pass it up to the PSTN?) I suspect we need an example. I have an X100P to the PSTN and an S100U to an Analog Phone (ext. 3100) Let's say my PSTN number is 555-1296. Usually, I have Asterisk take incoming calls on the X100P and send them to the analog phone on the S100U (exten = s,1,Goto(3100)) and from time to time I set Asterisk to call my cell phone, which it does over IAX with another asterisk server from a different PSTN number. What happens when another person calls me on 555-1296? I would like to know how to answer the call waiting both from the analog phone and from the cell phone. But the key is - needs to happen not at the asterisk level, but asterisk needs to pass the flash to the PSTN. In summary: I call from the analog phone over IAX. S100U in use, X100P not in use. - Call comes in, I can flash it to take the new call. I call from the analog phone over the PSTN. S100U in use, X100P in use. - Call comes in, I hear the PSTN-generated call waiting signal, but can't switch - Need help here. I get a call that is sent to my cell phone. S100U not in use, X100P in use. - Call comes in, I hear PSTN-generated call waiting signal on my Cell Phone, but can't switch - Need help on this one too. Any suggestions??? Thanks! -Matt ___ 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] Snom Intercom
rayers.list wrote: I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and trying to use the following dialplan: exten = 5100,1,SipAddHeader(Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris,,v) or exten = 5100,1,SipAddHeader(Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris) or exten = 5100,1,SetVar(_SIPADDHEADER=Call-Info: answer-after=0) exten = 5100,2,Dial(SIP/chris,,v) None of the above work. Can somebody tell me that this either is or is not possible? Ryan ___ 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 In the same vein, I would like the proper button to light when I call an extension. I have five extensions configured on the SNOM, 201-205. When I dial 203, as an example, the top button lights. I would like the third button, 203, to light on the incoming call. Thanks, ___ 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] Snom Intercom
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 09 August 2004 11:37 pm, Michael Welter wrote: In the same vein, I would like the proper button to light when I call an extension. I have five extensions configured on the SNOM, 201-205. When I dial 203, as an example, the top button lights. I would like the third button, 203, to light on the incoming call. Thanks, Yes, this would be good. I have contacted Snom about this situation. So far they have been very responsive to user requests. What you may consider doing is downloading their beta s/w (currently on 3.37). Then you can knock it around and let them know of any problems you see. I find it very good with very few annoyances. http://www.snom.com/download/share/ Make sure you get the one starting with snom200-version-SIP.bin if you use a 200, and so on. Release notes are on: http://www.snom.com/snom200_release_notes_en.php Modify your model number to match what you got. Remember it's Beta so some problems are to be expected. I prefer having the files on my local computer so I'm less dependent on reaching somone else if I need to revert back to an older version. So I got tftp setup and do my upgrades internally. YMMV. - -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD4DBQFBGEayljK16xgETzkRAjTwAKCri0ZphK9OPMnL9WRZZd8kLMtzXwCYzZeQ gelwyKsU1RWcOJ/noav2dA== =8uBe -END PGP SIGNATURE- ___ 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] asterisk with H.323 phone
On Mon, 9 Aug 2004 15:25:02 +0200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, I have 'Voipac NetPhone 210' phone apparatus with H.323 support Is there any way to connect it to asterisk? What exactly I need to do. Thank you very much. hi, Just google for oh323 and the first few results are ur answer. -- Manoj Kr. Gupta (MKG) Many ideas grow better when transplanted into another mind then the one where they sprung up. ___ 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] introduced Agents and * stops answering calls
Hi, I've looked through the list archives, bug tracker and cvs changelogs and can't see anything that refers to the particular problem i've seen recently. I'm running CVS-D2004.06.09.14.00.00-06/24/04-00:43:55 which I realise is not exactly recent, but I wanted to find out more before I updated. We've had asterisk running for our office using 7960's and a TE410P for a while, initially just doing what our old system did. The other day I introduced AgentCallbackLogin and assigned staff each an Agent id. Previously we'd just had the SIP/users specified in the queues.conf directly. Now I had the Agent/id specified. After a couple of hours we found the phones stopped ringing and asterisk stopped picking up the calls from the E1/PRI. Trying to see what was happening I would connect to the CLI and run show channels which would either return nothing except the CLI prompt or would not return at all. This happened a few times, a full restart fixed it each time. Has anyone else experienced this problem or have any suspicions to it's cause? I'd like to move forward to rc1 or the release when it happens... but I'm wary of having similar problems again... I've not been able to find any recognition that such a problem exists let alone that it has been fixed. Regards, - Sam -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) ___ 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] Click to Call
Hi, Ed Guy, (B (BDo you find interesting issue when getting CDR? (BIt has two billing leg on "click to call". One is origination leg and the (Bother is termination leg. After call drop, only termination leg CDR is (Blogged. (B (BRegards, (Byang (B (B (B-Original Message- (BFrom: [EMAIL PROTECTED] (B[mailto:[EMAIL PROTECTED] On Behalf Of Ed Guy (BSent: 2004$BG/(B8$B7n(B10$BF|(B 4:35 (BTo: [EMAIL PROTECTED] (BSubject: RE: [Asterisk-Users] Click to Call (B (BAndrei, (B (BIt is an activeX control and it no longer needs (Bto have the security level changed. (B (B(The earlier version was an unsigned applet that required the security (Blevel change until I bought the code signing certificate early this year.) (B (BIf you want to have a soft phone running in a web browser, (Byour approaches are somewhat limited and an ActiveX control (Bis probably the most expedient approach for the general user (Bpopulation. (read: very few people from this list ;-) ) (B (BSo, if you're Intranet can reach the Internet, connect your (BAsterisk server to FWD and simply have the users' click on (Byour icon and call your FWD account. This HTML code snippet (Bdoes it: (B (BA HREF="http://fwd.pulver.com/callme.php?userid=5" (BIMG SRC="http://fwd.pulver.com/myicon.php?userid=5" (BFWD# 5 (B/A (B (BChange it to use your userid, and you may replace the image as desired. (B (BIf you simply want click dial to dial, your choices are many... (BOn asterisk, I use the create a file in /var/spool/asterisk/outgoing (Bapproach with great success. (B (BHere's a simple shell script I use; it could easily be (Bmoved to php or a CGI after cleansing the parameters. (There must be (Bcomplete examples on voip-info.org) (B (B#!/bin/bash (B# (B# usage: callme fromnumber tonumber (B# (B (BPHONE=SIP/$1 (BDEST=$2 (B (BTMPFILE=/tmp/$$.call (B (Bcat $TMPFILE XXX (B# (BChannel: $PHONE (BMaxRetries: 2 (BRetryTime: 60 (BWaitTime: 30 (B# (BContext: internal-origination (BExtension: $DEST (BPriority: 1 (BXXX (B (Bmv $TMPFILE /var/spool/asterisk/outgoing/ (B# (B# end (B (BBest Regards, (BEd Guy @ pulver.com (B (B (B (B (B -Original Message- (B From: [EMAIL PROTECTED] (B [mailto:[EMAIL PROTECTED] Behalf Of Steven (B Critchfield (B Sent: Monday, August 09, 2004 2:44 PM (B To: [EMAIL PROTECTED] (B Subject: RE: [Asterisk-Users] Click to Call (B (B (B On Mon, 2004-08-09 at 13:26, Andrew Thompson wrote: (B Glen Hinkle wrote: (B Just write a CGI script that places a file in in the outgoing calls (B directory. /var/spool/asterisk/outgoing, I believe. This will (B accomplish what you're wanting. (B (B Did you even click the link? (B (B I saw in FWD site a phone on the web.. (click 612 link) (B http://www.freeworlddialup.com/advanced/beta_programs (B (B There it is again... Try it. (B (B As has been mentioned here, that app is an activeX control that appears (B to be locked to XP and ME, and worse yet is probably not capable of (B being used outside of IE. It even warns you that you have to lower your (B security settings to get it to work. (B (B Think four or five times about the inherent danger of having a app that (B requires you to reduce security settings on such a piss poor security (B wise app. As another carrot, I would laugh at any company that wanted my (B to do such a thing especially if they where a computer/technology (B company. It would get rediculed all over any public message board I saw. (B (B As a seperate option, the CGI solution above kind of gets a similar (B functionality. No it doesn't use the web browser, but it would allow you (B to collect a phone number and issue a call out to the person requesting (B the call. You then could select when and how to connect. (B (B (B -- (B Steven Critchfield [EMAIL PROTECTED] (B (B ___ (B Asterisk-Users mailing list (B [EMAIL PROTECTED] (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] truncated extensions
Greg Hill wrote:
On Sun, 8 Aug 2004, Kevin Johnson wrote:
I'm having a problem with extensions.
Any extension longer than 6 characters gets truncated to 6 characters.
For example,
exten = _7XX,3,NoOp(call for${EXTEN})
results in
call for 712345
when given
7123456
that's ${EXTEN}, not ${EXTEN:1}, right (I mean what's actually in your
extensions.conf file)? And you don't have any other extensions with the
'.' wildcard in them which might be getting matched instead? Next step:
mention which version you're running and maybe include extensions.conf.
Correct, ${EXTEN}.
I'm using CVS-HEAD-08/08/04-07:50:38
I've attached an extensions.conf file that exhibits the problem for me.
Sorry for the extra stuff in the file, but that's what I'm running,
minus some global variables not suitable for public consumption.
When dialing 8437624, I get the following output:
-- Executing NoOp(SIP/office1-b727, call for 843762 43762
6) in new stack
on the following line:
exten = _8.,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
... missing the final 4 in the dialed number.
I included a call to LEN, just to make sure I wasn't seeing things...
Oddly enough, the _4X dialplan doesn't exhibit the problem.
Any pointers would be greatly appreciated.
[general]
static=yes
writeprotect=yes
[globals]
OFFICE=sip/office1
KJJLAPTOP=iax2/kjjlaptop2
;;;
[default]
exten = s,1,Playback(all-your-base)
exten = s,2,Hangup
;include = thejohnsonpigs
[mainmenu]
exten = s,1,Wait,1
exten = s,2,Wait,1 ; wait 2 seconds to allow receipt of caller-id data
exten = s,3,Answer
exten = s,4,Playback(moo2)
exten = s,5,DigitTimeout,5
exten = s,6,ResponseTimeout,10
exten = s,7,Background(enter-ext-of-person)
exten = t,1,Playback(i-grow-bored)
exten = t,2,Hangup
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,5) ; loop on reprompting for extension
exten = 1,1,Macro(stdexten,1001,${OFFICE})
exten = 2,1,Macro(stdexten,1002,${KJJLAPTOP})
[voicemail]
exten = 4242,1,VoicemailMain
exten = 4242,2,Hangup
[thejohnsonpigs]
exten = 1001,1,Playback(transfer,skip)
exten = 1001,2,Macro(stdexten,1001,sip/office1)
exten = 1001,3,Congestion
exten = 1002,1,Playback(transfer,skip)
exten = 1002,2,Macro(stdexten,1002,IAX2/kjjlaptop2)
exten = 1002,3,Congestion
[outbound-to-iaxfwd-test]
exten = _4X,1,SetCallerId,${FWDNUM}
exten = _4X,2,SetCIDName,${FWDCIDNAME}
exten = _4X,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
exten = _4X,4,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},60,r)
exten = _4X,5,Playback(invalid)
exten = _4X,6,Hangup
exten = _4X,7,Congestion
exten = _8.,1,SetCallerId,${FWDNUM}
exten = _8.,2,SetCIDName,${FWDCIDNAME}
exten = _8.,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
;exten = _8.,4,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},60,r)
exten = _8.,4,NoOp(arf)
exten = _8.,5,Playback(invalid)
exten = _8.,6,Hangup
exten = _8.,7,Congestion
[inbound]
include = mainmenu
exten = _.,1,Goto(mainmenu,s,1)
exten = _.,2,Hangup
[inbound-from-sip]
include = default
[inbound-from-iax]
include = default
[inbound-from-iaxfwd]
include = inbound
[house]
;include = default
;include = thejohnsonpigs
include = outbound-to-iaxfwd-test
include = voicemail
;;;
[macro-stdexten]
exten = s,1,Background(one-moment-please)
exten = s,2,Dial(${ARG2},10)
exten = s,3,Voicemail(u${ARG1})
exten = s,4,Goto(default,s,1) ; If they press #, return to start
exten = s,103,Voicemail(b${ARG1})
exten = s,104,Goto(default,s,1); If they press #, return to start
;;;
[Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs
Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse, Nufone, broadvoice. ___ 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] 831 Santa Cruz/Watsoncille, Calif. DIDs
gafachi This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lists-jmhunter Sent: Tuesday, August 10, 2004 1:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse, Nufone, broadvoice. ___ 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 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
