Re: [asterisk-users] SIP <> GSM
On Jan 20, 2008 4:40 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > I'd like to add a device to my Asterisk server to leverage my cellular > account. Does anyone on-list have experience with hardware gateways vs > using cah_bluetooth and an old cell phone? We use the Junghanns.NET duoGSM PCI card with the bristuff driver. Did not have any problems with it yet, works as expected. Regards, Geert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box?
Hello Steve, On Nov 26, 2007 4:06 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > I know it is a strange arrangement but due to contracts, it is what it > is, no PRI for now. > > I wonder if anyone on the list has run a server with both types of cards > installed? Results? > Not quite the same, but we run the Junghanns quadbri card together with a TDM40b and the Junghanns GSM card. Works without problems (after making sure the modules are always loaded in the right order). No Sangoma hardware though. Regards, Geert ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you handle NAT?
On 6/28/05, Sebastian Silva <[EMAIL PROTECTED]> wrote: > Hi everyone. > > 1. Asterisk as a SIP client behind nat, connecting to outside SIP Proxies: > #1 works with a NAT-supporting proxy as SIP Express router as the > outside proxy. (Get an account at IPtel.org and try!). Fails with Free > World Dialup. > > 2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies: > #2 Works- no NAT in between > > 3. Asterisk as a SIP server behind nat, clients on the outside > connecting to Asterisk: > #3 Works with port forwarding and some header mangling magic > > 4. Asterisk as a SIP server behind nat, clients on the inside connecting > to Asterisk: > #4 Works - no NAT in between > > 5. Asterisk as a SIP client outside nat, connecting to outside SIP proxies: > #5 is no problem. No NAT in the middle > > 6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies: > #6 is a problem if no port forwarding is done, similar to 3 above. > > 7. Asterisk as a SIP server outside nat, clients on the outside > connecting to Asterisk: > #7 is no problem. No NAT in the middle > > 8. Asterisk as a SIP server outside nat, clients on the inside > connecting to Asterisk: > #8 is solved with nat=yes and qualify=xxx in sip.conf for the client in > most cases. Some clients (X-lite) assist themselves by using STUN and > sending UDP keep-alive packets. Qualify sends keep-alive packets from > Asterisk to the client on the inside. > > from wiki > > Now, if you net to define a NAT, you have to set asterisk to > "canreinvite=no", "qualify=yes" and "nat=1". > > Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server > you should set asterisk to "canreinvite=no", "qualify=no" and "nat=0" > (the STUN configuration is in your agents). > You can use STUN instead of nat=yes (if the phone supports STUN properly). However, our experience is that we also need qualify=yes to prevent the phones becoming unreachable. Geert > hank wrote: > > how easy is it to set up a stun server? with asterisk amd will this fix > > part of the nat problem? > > - Original Message - From: "Ray Van Dolson" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Tuesday, June 28, 2005 8:14 AM > > Subject: Re: [Asterisk-Users] How do you handle NAT? > > > > > >> We've been feeling our way along with the NAT stuff (using SIP) as well. > >> > >> At this point we are fairly small, so the keep-alive packets are not > >> too bad. > >> What type of user load are you at and what are the specs on your > >> Asterisk box? > >> I'm concerned we may run into this as well. > >> > >> We do have the luxury that each Sipura device we use is sitting behind > >> its own > >> NAT (a customer CPE). So we can do port-forwarding and in combination > >> with a > >> STUN server (MyStun), things work quite well. The only issues left to > >> deal > >> with are a lingering problem with ip_conntrack entries staying cached > >> because > >> of the "keep alive" packets due to qualify=yes after the CPE's IP address > >> changes. > >> > >> Curious to hear other's setups as well. I would *love* to start using > >> the > >> IAXy instead, but it has a couple shortcomings over the Sipura 2002's > >> we're > >> using now: > >> > >> - About $10/more > >> - Only has one line (apparently two lines is a bit more of a selling > >> point). > >> > >> Still trying to figure out a good way to make a case for the IAXy though. > >> > >> Ray > >> > >> On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote: > >> > >>> We are interested in how other people are handling NAT problems. We have > >>> several customers all of which have some sort of firewall/NAT device at > >>> their location. For simplicity sake, all customers' internal networks > >>> are 192.168.*.*. > >>> > >>> Our asterisk box is on public IP not blocked by any FW/NAT. > >>> > >>> I use QUALIFY=yes on all our customers' phones and I feel that sending > >>> out 80-something keep-alive packets is causing our box to crawl and > >>> cause bad calls. > >>> > >>> Would SER be better in this case? Should I have phones register with SER > >>> instead of with Asterisk? > >>> > >>> Thanks, > >>> Matthew > >>> > >>> P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in > >>> other real world, working, solutions. > >> > >> ___ > >> Asterisk-Users mailing list > >> Asterisk-Users@lists.digium.com > >> 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 > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Sebastian Silva > G R U P O G
Re: [Asterisk-Users] sip device discussion and reviews (Snom 190 request)
Simon Dorfman wrote: I'd love to hear a review of any Snom Phones. I'm waiting for the Snom 190 before I buy my first hardware VoIP phone. It's supposed to be around $150 or less. I've already read what voip-info has to say: http://voip-info.org/tiki-index.php?page=Snom%20Phones http://voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom ...as well as searched this lists archives, but I'd still like to hear more feedback about Snom phones. We use the SNOM's. They are excellent, their support is excellent and the development of new features in the firmware is very fast. I have one major gripe about them, the speaker is not good enough for long conversations. Regards, Geert ___ 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: Snom200 ?
nicolas wrote: Snom is a great phone, especially in conjunction with Asterisk. See for more information and tips and tricks the following URL: http://www.voip-info.org/wiki-SNOM+phones This is because i bought a snom. In what context? If the SNOM is reply-ing this when idle, this is not normal behaviour. It is however possible that you are running an old firmware, where the busy setting status is not displayed on the phone. Try to upgrade. This behaviour effects if i dialing the snom (in idle) without making an "exten => s,x,Answer", witch is described in several asterisk docs. You mean you are doing something like: exten => 666,Dial(SIP/snom) or are you using the SNOM as stand alone phone (dial directly to it's IP address)? And the call indication function even do not work with the snom200. You mean the Call Waiting Indication? The firmware is up to date (2.05b) also tested it with 2.04g,t. Regards, Geert Geert Nijpels wrote: nicolas wrote: Hi all, Got SIP response 486 "Busy Here" back from x.x.x.x I become this message if a call is coming in and i have read this is normal with snom. In what context? If the SNOM is reply-ing this when idle, this is not normal behaviour. It is however possible that you are running an old firmware, where the busy setting status is not displayed on the phone. Try to upgrade. If it is so, then a right extension.conf is unuseable and snom be not the right phone. Snom is a great phone, especially in conjunction with Asterisk. See for more information and tips and tricks the following URL: http://www.voip-info.org/wiki-SNOM+phones please help. nicolas Regards, Geert ___ 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] Snom200 ?
nicolas wrote: Hi all, Got SIP response 486 "Busy Here" back from x.x.x.x I become this message if a call is coming in and i have read this is normal with snom. In what context? If the SNOM is reply-ing this when idle, this is not normal behaviour. It is however possible that you are running an old firmware, where the busy setting status is not displayed on the phone. Try to upgrade. If it is so, then a right extension.conf is unuseable and snom be not the right phone. Snom is a great phone, especially in conjunction with Asterisk. See for more information and tips and tricks the following URL: http://www.voip-info.org/wiki-SNOM+phones please help. nicolas Regards, Geert ___ 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 200
Hermann Wecke wrote: Sorry to ask this here but I believe that it is the best place to receive a feedback... I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *, and the overall impression about these phones... _ We use the SNOM with * and they work good. We worked a lot with the excellent SNOM support to resolve MWI issues, and now MWI works like a charm. Especially their NAT features are very good, which I can not say about the Cisco 7960's. Also, they have IPv6 on their roadmap, and Cisco confirmed they will NOT do this with their 7960. Regards, Geert ___ 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] MWI indicator on SNOM200 doesn't disappear
Ian White wrote: On Apr 22, 2004, at 23:48, Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. "Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied?" I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: "application/simple-message-summary", but with "Content-Type: text/plain", so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) Any idea what sort of time frame before chan_sip2 becomes usable in a production environment, or at least becomes part of the CVS tree? I see your note saying that you are using it in production. I'm using it with some changes with -stable. It's developed by oej for -devel. Works great with my SNOM's and Cisco 9760. You can get chan_sip2 through the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=759 I can also send you my -stable version, but you can backport it with some minor trouble yourself. Geert ___ 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] MWI indicator on SNOM200 doesn't disappear
Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. "Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied?" I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: "application/simple-message-summary", but with "Content-Type: text/plain", so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Kind regards, Geert ___ 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] SER Asterisk problem
Welesley Sibelson Dias wrote:
Hi All.
I'am using Asterisk with SER. I can make call between two internal VoIP
gateways or from na internal to external VoIP gateway. But when I get a
external call, this call hang ups 5 seconds after and I reveive the
following messages
*CLI> -- Executing Dial("SIP/16008-3d17",
"SIP/16007&SIP/16006|20|tr") in new stack
-- Called 16007
-- Called 16006
-- SIP/16007-8c24 is ringing
-- SIP/16007-8c24 answered SIP/16008-3d17
-- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar
30 13:53:11 WARNING[1125685952]: chan_sip.c:495
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102
(Request)
Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102
(Request)
Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 8 (Response)
=3D=3D Spawn extension (sip, 1000, 1) exited non-zero on
'SIP/16008-3d17'
Jadylson da Rocha Passos Bomfim
I know of a GrandStream bug which generates a wrong ack to the 200 OK
asterisk sends on connecting. SER drops this ack and asterisk drops the
call, as it should. This is fixed in latest firmware image.
Kind regards,
Geert
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Re: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE
John F. Baird wrote: From the Cisco docs on the 7960G: Features Q. What inline power standard is supported? A. Cisco Inline Power, a pre-standard implementation, is supported at initial release. IEEE inline power will be supported in the future. The original IEEE 802.3 standard did not include specifications for power over Ethernet, which is now being addressed in the IEEE 802.3af draft. The draft should be ratified by mid-2003. Cisco is actively involved in the 802.3af taskforce and is committed to the development and support of the IEEE inline power standard. To preserve customer's existing investment, Cisco Inline Power will continue to be supported. Q. What local power option available for the Cisco IP Phone 7905G? A. The local power option is provided through a 48 VDC Cisco IP Phone power supply, order code, "CP-PWR-CUBE". With a Foundry POE switch using mode "inline power legacy-powerdevice" the Cisco 7960G works. Geert ___ 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 200
Barry Fawthrop wrote: Progress It seems I can't hear the Say Time, due to RTP Double NAT I'm guess this is both problem 1 and 2 really issue. My config: IP Phone <-> Router (Nat) <-> Internet <-> Linux (NAT) <-> * Server ANyone know of work arounds the double NAT? or other methods to route RTP with snom 200s, to work around this? I think you can make progress with the following link: http://www.voip-info.org/tiki-index.php?page=NAT%20and%20VOIP Have fun, Geert ___ 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 200
Barry Fawthrop wrote: Thanks Willy and others It seems I am not able to make myself clear on my two problems I wish to try again, as I'm sure it is not the phones, but some stupid config problem on my part. I repeat alot of what I have said in order to try make myself clear. That is all problem 1, the phones report busy, while sitting idle This is a known problem. Sometimes the SNOM's seem to go to BUSY without any cause. At least no indicator is shown in the display. A possible solution is posted here: http://www.voip-info.org/wiki-SNOM+phones If it does not solve the problem, try to reset the phone to default settings and power cycle it. With the latest firmware I did not see the problem yet, but I see the other bugs (crash + transfer, I'm busy emailing with SNOM about these bugs). Problem 2, If I pick up the handset I hear the dialtone (proof the phone is connected) When I dial an extension which is set to play the time and date, the * CLI scrolls the voice saying date an time. Yet the Handset is silent, Why? If I hear the dialtone at the start why does the handset go dead, surely I should hear the voice on the other side talking (in this case the * server)? The dialtone is no indication you can setup an RTP stream. You should test it with the asterisk built in ECHO server. Make sure there is no firewall activated which can block the traffic. If it still doesn't work, check the "sip debug" output for errors or retransmits. Kind regards, Geert ___ 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] usb-uhci -- where to find it?
Darrin Johnson wrote: Hello all, I do not have a Zaptel card, but still wanted to utilize the conferencing capabilities in Asterisk. I am trying to find a site to download the required usb-uhci module, but have not had much luck. Can anyone point me in the right direction? Thanks, Darrin Johnson Systems Engineer IS Domain Inc. You must checkout zaptel cvs for that. Edit the makefile and install to enable ztdummy. http://www.asterisk.org/index.php?menu=download Geert ___ 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] Dual Xeon
reseaux wrote: Dear Geert I use * with 1 TE400P on Dual Xeon with 1GByte of RAM HT everyday with little 30 channels load of calls at time, can you give me more info about problem in this kind of configuration? thanks Dimitri I never did experience problems that could be directly linked to HT. However, I was told at #asterisk HT would not give much of a performance gain and can cause problems with sound quality. Also I had a problem with calls having 3 out of 5 calls no sound while the RTP stream did build up correctly, this problem went away after disabling HT with my Xeon proc, but unfortunately I also changed other things in the hardware configuration so I can not point it to the HT stuff. I'm sure that it wasn't a configuration error and also that the memory is working correctly (memtest). Kind regards, Geert ___ 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] Dual Xeon
Ed Devine wrote: When compiling Asterisk for a dual XEON based system are there any caveats or "switches" that we need to be aware of? Well, for zaptel hardware you need to uncomment the SMP entry in the zaptel Makefile. Also I would turn off Hyperthreading (in the bios). It may cause problems. Kind regards, Geert
Re: [Asterisk-Users] Snom 100 Code Recommendation
Jason Ross wrote: G'Day, I've got a Snom 100 and am running the 2.03o SIP code. Basically I'm having DMTF problems no matter what configuration I try. And as yet I haven't downgraded it to see if an earlier release makes a difference Just wondering if anyone can provide some guidance as to what the best release of code for this phone may be. I've been having "busy" problems with the 2.03x firmware versions, but no DTMF problems. I configured the phone for DTMF outband, with asterisk configured as dtmfmode=rfc2833. I'm running 2.02z. Check also: http://www.voip-info.org/wiki-SNOM+phones Kind regards, Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP debug logs
Steve Foy wrote: This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! I dont know if it's possible using asterisk. You can use the command 'script -a ' that will record everything at the prompt. Geert ___ 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] Problems with chan_sip: random calls have no sound withouth any errors
Eric Wieling wrote: Asterisk is still saying it accepts G729. That is prolly the problem. Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1. If there any reason you are allowing both ulaw AND alaw. Sorry forgot to mention it. I'm already at latest CVS, but I have this problem also with 0.7.1. Well I use alaw and ulaw because all my phones support these codecs. But I get this problem with other codec configurations too. Kind regards, Geert On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote: Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work best. I could also test from another phone with the same results. The S is for Success (can talk), the F is for Failure(Call gets setup but no speech/sound). Cisco 7960 to SNOM S,S,F,S,F,F,F,S,S,S,S,S,F Cisco 7960 to GS S,F,S,S,F,S,S,F,S,F, I placed a sip debug from asterisk for each situation at the following URL: http://audix.noc.ams-ix.net/asterisk/dumps/ - cisco_to_gs_failure.txt - cisco_to_gs_success.txt - cisco_to_snom_success.txt - cisco_to_snom_failure.txt Somebody have a clue? I'm thinking of filing a bug but I want to make sure this is no configuration or other problem at my side. Thanks and kind regards, Geert Nijpels ___ 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] Problems with chan_sip: random calls have no sound withouth any errors
Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work best. I could also test from another phone with the same results. The S is for Success (can talk), the F is for Failure(Call gets setup but no speech/sound). Cisco 7960 to SNOM S,S,F,S,F,F,F,S,S,S,S,S,F Cisco 7960 to GS S,F,S,S,F,S,S,F,S,F, I placed a sip debug from asterisk for each situation at the following URL: http://audix.noc.ams-ix.net/asterisk/dumps/ - cisco_to_gs_failure.txt - cisco_to_gs_success.txt - cisco_to_snom_success.txt - cisco_to_snom_failure.txt Somebody have a clue? I'm thinking of filing a bug but I want to make sure this is no configuration or other problem at my side. Thanks and kind regards, Geert Nijpels ___ 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] Multiple Line Appearances
Jonathan Moore wrote: I may be misunderstanding the question but what is wrong with using an extension line like the following? exten => 3,1,Dial(SIP/snom200&SIP/snom100&SIP/gs1,15,Ttr) I use this to have all three desk phones in my office ring on the same menu option/extension. Yes, but if you are mobile and want to get called at your number, it would be a good thing you can log in anywhere you want and have the calls forked to both places (or the one with priority). Also the case with a phone at home and at work. Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP URI matching
Geert Nijpels wrote: Hi, Can somebody tell me how to do SIP uri matching? I have in my [from-sip] context a few extensions included, this works. So I can call: [EMAIL PROTECTED] But I also want to be able to call my sip [peer] or user name. Is this possible? As in: [EMAIL PROTECTED] For the archives: It already works: [from-sip] exten => geert2,1,Dial(SIP/geert2) Thanx jtodd! Geert ___ 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] Multiple Line Appearances
MLS Drop for SysAdmin wrote: Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for SIP extensions. Thanks for your feedback Sam Zener I'm not totally sure what you want, but I'll give it a shot:) Multiple SIP registrations are not (yet??) possible, even though this is described in the RFC. Also see bug 157. Hopefully this will be implemented in chan_sip2, this is bug nr. 759. What you can do is: Dial(technology/extension&technology2/extension2&technology3/extension3) or code it yourself (or put a money bounty on it). Kind regards, Geert ___ 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] dialing wrong numbers
Jon Pounder wrote: hi, for some unexplained reason, I have trouble dialing "2" on certain phones. (it gets picked up as something else sometimes) not sure why this is, and if the phone, channel bank, or software are to blame. I just try to ignore it. If this is a Grandstream phone, you could try if a firmware upgrade helps: http://www.voip-info.org/tiki-index.php?page=Budgetone Kind regards, Geert ___ 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 URI matching
Hi,
Can somebody tell me how to do SIP uri matching?
I have in my [from-sip] context a few extensions included, this works.
So I can call:
[EMAIL PROTECTED]
But I also want to be able to call my sip [peer] or user name. Is this
possible? As in:
[EMAIL PROTECTED]
Also, what do you guys do for outbound sip calls? Currently I have a
prefix for it:
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Kind regards,
Geert
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Re: [Asterisk-Users] Snom 200 phones not working.
Ariel Batista wrote: I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration files. I have followed the Snom FAQ step by step. But it's still not working. I had the same problem, so I emailed SNOM. After a quick and clear reaction from SNOM, the following turns out: I have an SRV record set for Asterisk using both TCP and UDP, because I was first experimenting with SER and that SIP proxy DOES support TCP. Asterisk does not. So in firmware 2.02t the phone tried udp automatically, and in firmware 2.03o the phone tried to use tcp, which will not work in Asterisk. Removing the TCP SRV entry solved my problem Maybe this will solve your problem. However, I still have other problems with the SNOM phones: - All sound stops working sometimes (also ringtone) - Speech sometimes not working (not sure if it is RTP problem or SNOM firmware problem) - Sometimes the phone returns BUSY when not busy in firmware 2.02t. Resetting the phone or adding a new SIP line solves this. In firmware 2.03o, this is different. The phone does not respond and Asterisk gives the error: phone CIRCUIT BUSY. I have seen other posts here from people having the same problems. Other people having the same problems? It makes my case a bit more clear at SNOM when I can point them to a thread with a lot of people having the same problems:-) I am emailing with SNOM about these issues. Kind regards, Geert Nijpels ___ 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
